
<--- SIP read from UDP:172.22.11.155:5060 --->
INVITE sip:0935377349@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK72884e55;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 00:30:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 680507915 680507915 IN IP4 172.22.11.155
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.155
t=0 0
m=audio 17332 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.155:5060 (NAT)
Using INVITE request as basis request - 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
peer=(null).
No matching peer for '805-172.22.11.155' from '172.22.11.155:5060'
user_and_ip=805-172.22.11.155, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.155.
Found peer '172.22.11.155' for '805' from 172.22.11.155:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.155:17332
Looking for 0935377349 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:805@172.22.11.155:5060>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK72884e55;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK72884e55;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18200
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK72884e55;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>;tag=as43d03dae
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 88871437 88871437 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18200 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18200
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK72884e55;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>;tag=as43d03dae
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 88871437 88871438 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18200 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.155:5060 --->
ACK sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK361d0632;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>;tag=as43d03dae
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.155:5060 --->
BYE sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK5d425529;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>;tag=as43d03dae
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.155:5060 (no NAT)
Scheduling destruction of SIP dialog '5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK5d425529;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as703e8a0d
To: <sip:0935377349@172.22.11.181>;tag=as43d03dae
Call-ID: 5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '5f91545c509fb0982b0c4a3707245fc8@172.22.11.155:5060' Method: BYE

<--- SIP read from UDP:172.22.11.155:5060 --->
INVITE sip:0935377349@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK428a7769;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 00:52:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1922622281 1922622281 IN IP4 172.22.11.155
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.155
t=0 0
m=audio 10784 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.155:5060 (NAT)
Using INVITE request as basis request - 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
peer=(null).
No matching peer for '805-172.22.11.155' from '172.22.11.155:5060'
user_and_ip=805-172.22.11.155, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.155.
Found peer '172.22.11.155' for '805' from 172.22.11.155:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.155:10784
Looking for 0935377349 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:805@172.22.11.155:5060>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK428a7769;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK428a7769;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12166
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK428a7769;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>;tag=as056ad864
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1237150610 1237150610 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12166 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12166
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK428a7769;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>;tag=as056ad864
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1237150610 1237150611 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12166 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.155:5060 --->
ACK sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK4d7a4c19;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>;tag=as056ad864
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.155:5060 --->
BYE sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK58171bdd;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>;tag=as056ad864
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.155:5060 (no NAT)
Scheduling destruction of SIP dialog '1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK58171bdd;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as1eac9b01
To: <sip:0935377349@172.22.11.181>;tag=as056ad864
Call-ID: 1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
Really destroying SIP dialog '1116b0722dbe8ad54d17d88f46834654@172.22.11.155:5060' Method: BYE

<--- SIP read from UDP:172.22.11.155:5060 --->
INVITE sip:0935377349@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK0c81c3f5;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 00:53:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 45556218 45556218 IN IP4 172.22.11.155
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.155
t=0 0
m=audio 16916 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.155:5060 (NAT)
Using INVITE request as basis request - 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
peer=(null).
No matching peer for '805-172.22.11.155' from '172.22.11.155:5060'
user_and_ip=805-172.22.11.155, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.155.
Found peer '172.22.11.155' for '805' from 172.22.11.155:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.155:16916
Looking for 0935377349 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:805@172.22.11.155:5060>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK0c81c3f5;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK0c81c3f5;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14312
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK0c81c3f5;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>;tag=as70a6ff85
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 831685004 831685004 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14312 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14312
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK0c81c3f5;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>;tag=as70a6ff85
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935377349@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 831685004 831685005 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14312 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.155:5060 --->
ACK sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK1f7703dc;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>;tag=as70a6ff85
Contact: <sip:805@172.22.11.155:5060>
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.155:5060 --->
BYE sip:0935377349@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK5af38410;rport
Max-Forwards: 70
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>;tag=as70a6ff85
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.155:5060 (no NAT)
Scheduling destruction of SIP dialog '7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.155:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.155:5060;branch=z9hG4bK5af38410;received=172.22.11.155;rport=5060
From: "Vladislav Maslennikov" <sip:805@172.22.11.155>;tag=as6e260665
To: <sip:0935377349@172.22.11.181>;tag=as70a6ff85
Call-ID: 7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '7588edd4423ce62040e4659e30e1aca1@172.22.11.155:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK640c2e42;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:34:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1179406233 1179406233 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14800 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14800
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK640c2e42;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK640c2e42;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17566
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK640c2e42;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>;tag=as61b8b7c9
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1192342488 1192342488 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17566 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 17566
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK640c2e42;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>;tag=as61b8b7c9
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1192342488 1192342489 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17566 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK290eeab5;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>;tag=as61b8b7c9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6bf4e3e3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>;tag=as61b8b7c9
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6bf4e3e3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c47d442
To: <sip:0935288746@172.22.11.181>;tag=as61b8b7c9
Call-ID: 4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK340c5f96;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:35:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 92531632 92531632 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14570 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14570
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK340c5f96;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK340c5f96;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12308
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK340c5f96;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>;tag=as6c6a350c
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 427859240 427859240 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12308 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12308
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK340c5f96;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>;tag=as6c6a350c
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 427859240 427859241 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12308 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4d74121d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>;tag=as6c6a350c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4503eb880d46dcd35d08ad646cb9af2b@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6a01eaf9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>;tag=as6c6a350c
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '6137769535a32b203de34a3b757f8124@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6a01eaf9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052079ba
To: <sip:0935288746@172.22.11.181>;tag=as6c6a350c
Call-ID: 6137769535a32b203de34a3b757f8124@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '6137769535a32b203de34a3b757f8124@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK426ab1db;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:44:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 642150864 642150864 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18154 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18154
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK426ab1db;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK426ab1db;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11006
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK426ab1db;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>;tag=as21304534
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 851741874 851741874 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11006
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK426ab1db;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>;tag=as21304534
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 851741874 851741875 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b3161ad;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
To: <sip:0939011286@172.22.11.181>;tag=as21304534
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5ae2f99e;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as21304534
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5ae2f99e;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as21304534
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0da35c73
Call-ID: 4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4ca78f1370edb2ca1b576cc134e750c1@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51def5a2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:48:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2043878222 2043878222 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11078 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11078
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51def5a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51def5a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18344
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51def5a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>;tag=as34ef7410
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 744026351 744026351 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18344 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18344
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51def5a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>;tag=as34ef7410
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 744026351 744026352 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18344 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK78247973;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>;tag=as34ef7410
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931889554@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:48:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1264940623 1264940623 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13162 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13162
Looking for 0931889554 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17372
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>;tag=as44c9ccd0
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1993879964 1993879964 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17372 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48af5fe0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>;tag=as34ef7410
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48af5fe0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2563f177
To: <sip:0939319680@172.22.11.181>;tag=as34ef7410
Call-ID: 376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>;tag=as44c9ccd0
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931889554@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fff707e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as321ee980
To: <sip:0931889554@172.22.11.181>;tag=as44c9ccd0
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931889554@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK110714b2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:49:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1585385220 1585385220 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14708 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14708
Looking for 0931889554 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK110714b2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK110714b2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17326
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK110714b2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>;tag=as1305d60a
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 410497849 410497849 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17326 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '376e31ac77047eaa1dd95a6b672e2e71@172.22.11.142:5060' Method: BYE
Audio is at 17326
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK110714b2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>;tag=as1305d60a
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 410497849 410497850 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17326 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931889554@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK173b5926;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
To: <sip:0931889554@172.22.11.181>;tag=as1305d60a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0e10f346176099df2832b8fa4f7f4f00@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK026413e9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:50:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1205221744 1205221744 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19786 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19786
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK026413e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK026413e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10670
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK026413e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>;tag=as7a646dfc
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 197930211 197930211 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10670 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK40f119e5;rport
Max-Forwards: 70
From: <sip:0931889554@172.22.11.181>;tag=as1305d60a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK40f119e5;received=172.22.11.181;rport=5060
From: <sip:0931889554@172.22.11.181>;tag=as1305d60a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3e34b8ad
Call-ID: 6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '6494daff44bedeac3c93be2f596c639a@172.22.11.142:5060' Method: ACK
Audio is at 10670
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK026413e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>;tag=as7a646dfc
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 197930211 197930212 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10670 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08e256c3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
To: <sip:0939319680@172.22.11.181>;tag=as7a646dfc
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5b65a333;rport
Max-Forwards: 70
From: <sip:0939319680@172.22.11.181>;tag=as7a646dfc
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5b65a333;received=172.22.11.181;rport=5060
From: <sip:0939319680@172.22.11.181>;tag=as7a646dfc
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7a23f895
Call-ID: 0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '0f25e86c7a3febbb2c7d8ebc535050d0@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27915112;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:56:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 37414160 37414160 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13848 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13848
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27915112;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27915112;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13608
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27915112;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>;tag=as54cb172b
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1323816790 1323816790 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13608 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13608
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27915112;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>;tag=as54cb172b
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1323816790 1323816791 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13608 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ba05703;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>;tag=as54cb172b
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e1451d9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>;tag=as54cb172b
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e1451d9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as108465a7
To: <sip:0939011286@172.22.11.181>;tag=as54cb172b
Call-ID: 7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '7fc05cd215144ac92b0195c9155486d0@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0933027914@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK03fe3427;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:58:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 249962164 249962164 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13472 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13472
Looking for 0933027914 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK03fe3427;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933027914@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK03fe3427;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933027914@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13622
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK03fe3427;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>;tag=as4e3ff92b
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933027914@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 586917268 586917268 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13622 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13622
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK03fe3427;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>;tag=as4e3ff92b
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933027914@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 586917268 586917269 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13622 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0933027914@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1e4ad7be;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
To: <sip:0933027914@172.22.11.181>;tag=as4e3ff92b
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK54b2b4f1;rport
Max-Forwards: 70
From: <sip:0933027914@172.22.11.181>;tag=as4e3ff92b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK54b2b4f1;received=172.22.11.181;rport=5060
From: <sip:0933027914@172.22.11.181>;tag=as4e3ff92b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as037008ca
Call-ID: 40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '40cbc0331c7c019915c7bd8a7d297cd2@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:58:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 574228214 574228214 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14150 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14150
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17884
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>;tag=as4c3deb13
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1871894669 1871894669 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17884 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931889554@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33e61217;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:58:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1168795475 1168795475 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13388 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13388
Looking for 0931889554 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33e61217;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33e61217;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14952
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33e61217;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>;tag=as056f615c
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 254413832 254413832 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14952 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>;tag=as4c3deb13
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4f2d6735;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cec1c5f
To: <sip:0939319680@172.22.11.181>;tag=as4c3deb13
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:58:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 753225683 753225683 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18026 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18026
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15892
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>;tag=as5160cd2a
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 221692480 221692480 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15892 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14952
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33e61217;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>;tag=as056f615c
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931889554@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 254413832 254413833 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14952 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931889554@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b0feac;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
To: <sip:0931889554@172.22.11.181>;tag=as056f615c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '43393859660b8b11339689ad09e26d2a@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>;tag=as5160cd2a
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK02d65c72;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46343daf
To: <sip:0939319680@172.22.11.181>;tag=as5160cd2a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 43393859660b8b11339689ad09e26d2a@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:59:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1492584734 1492584734 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15648 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15648
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15022
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>;tag=as64e3fbb3
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 546252111 546252111 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15022 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '3303a1a158c69e577998220e54a63e97@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK739ac503;rport
Max-Forwards: 70
From: <sip:0931889554@172.22.11.181>;tag=as056f615c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK739ac503;received=172.22.11.181;rport=5060
From: <sip:0931889554@172.22.11.181>;tag=as056f615c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as49aca8d1
Call-ID: 3303a1a158c69e577998220e54a63e97@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3303a1a158c69e577998220e54a63e97@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>;tag=as64e3fbb3
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7e46b98f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as31e03102
To: <sip:0939319680@172.22.11.181>;tag=as64e3fbb3
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:59:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1290407235 1290407235 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18386 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18386
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11912
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>;tag=as01b53c31
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 935587851 935587851 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11912 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '24de169a0e9065f87e13eb0903c20c50@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>;tag=as01b53c31
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK108f93af;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as66d2bb61
To: <sip:0939319680@172.22.11.181>;tag=as01b53c31
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:59:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 960100867 960100867 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15688 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15688
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10058
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>;tag=as27a7b312
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1492111699 1492111699 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10058 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '43393859660b8b11339689ad09e26d2a@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>;tag=as27a7b312
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>;tag=as27a7b312
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06a431e9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56e65a95
To: <sip:0939319680@172.22.11.181>;tag=as27a7b312
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '35bdfa6d06cba1944ea7d5dd75853e1d@172.22.11.142:5060' Method: ACK
Really destroying SIP dialog '5e15f8050ae08f0a66dd515c7690a80d@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK279a5f54;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 05:59:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 459459153 459459153 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10254 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10254
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK279a5f54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK279a5f54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17192
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK279a5f54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>;tag=as372e66ac
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1036580615 1036580615 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17192 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '20c324865afb000d73a04d3d59a3871b@172.22.11.142:5060' Method: ACK
Audio is at 17192
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK279a5f54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>;tag=as372e66ac
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1036580615 1036580616 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17192 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK510425a6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
To: <sip:0939319680@172.22.11.181>;tag=as372e66ac
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK07c67285;rport
Max-Forwards: 70
From: <sip:0939319680@172.22.11.181>;tag=as372e66ac
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK07c67285;received=172.22.11.181;rport=5060
From: <sip:0939319680@172.22.11.181>;tag=as372e66ac
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5bc79d0b
Call-ID: 413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '413e966f66ee2e205891a34e0580eab3@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0631515468@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK26651456;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:00:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2137645824 2137645824 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13134 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13134
Looking for 0631515468 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK26651456;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631515468@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK26651456;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631515468@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10468
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK26651456;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>;tag=as4459da0e
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631515468@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 17326893 17326893 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10468 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10468
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK26651456;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>;tag=as4459da0e
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631515468@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 17326893 17326894 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10468 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0631515468@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2a2b0711;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>;tag=as4459da0e
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0631515468@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK222195d9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>;tag=as4459da0e
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK222195d9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as16b8b590
To: <sip:0631515468@172.22.11.181>;tag=as4459da0e
Call-ID: 70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '70fb5ae860ee7ef70196bae8685cb444@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931661267@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51cabe8d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:07:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 844573124 844573124 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18504 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18504
Looking for 0931661267 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51cabe8d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51cabe8d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13948
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51cabe8d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>;tag=as6d65010d
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1324439926 1324439926 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13948 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13948
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK51cabe8d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>;tag=as6d65010d
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1324439926 1324439927 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13948 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931661267@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48291faf;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>;tag=as6d65010d
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:08:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 180671150 180671150 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19208 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19208
Looking for 0637990999 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10268
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>;tag=as2a98b15e
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 62565708 62565708 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10268 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0931661267@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK21e69985;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>;tag=as6d65010d
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK21e69985;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2e89118d
To: <sip:0931661267@172.22.11.181>;tag=as6d65010d
Call-ID: 1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '1a4e65ff468399e65585c2d95d66c53b@172.22.11.142:5060' Method: BYE
Scheduling destruction of SIP dialog '4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>;tag=as2a98b15e
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637990999@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK69ebce45;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4cb72ac4
To: <sip:0637990999@172.22.11.181>;tag=as2a98b15e
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4c1094f61f925ef003ec219d0a11272f@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fdef82e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:22:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1299473415 1299473415 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13368 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13368
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fdef82e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fdef82e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10268
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fdef82e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>;tag=as5ed2d3e4
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 452097870 452097870 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10268 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10268
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fdef82e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>;tag=as5ed2d3e4
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 452097870 452097871 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10268 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK13544070;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>;tag=as5ed2d3e4
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23e835b7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>;tag=as5ed2d3e4
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23e835b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5b49bd43
To: <sip:0638708675@172.22.11.181>;tag=as5ed2d3e4
Call-ID: 4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '4110be9a19f695d03dc9cb0958820109@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:36:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 957426326 957426326 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12214 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 205be52e1423944365b48e915c438f48@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12214
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11586
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>;tag=as27e5ad31
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 839443769 839443769 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11586 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '205be52e1423944365b48e915c438f48@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>;tag=as27e5ad31
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05e64edc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as12b2af55
To: <sip:0638708675@172.22.11.181>;tag=as27e5ad31
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 205be52e1423944365b48e915c438f48@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:36:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1742723962 1742723962 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15146 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15146
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19290
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>;tag=as08a08c7b
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1039890632 1039890632 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19290 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>;tag=as08a08c7b
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>;tag=as08a08c7b
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6960d5d1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4840447d
To: <sip:0638708675@172.22.11.181>;tag=as08a08c7b
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3d2e8bd63a904f62230aae78042e6122@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:36:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 867152707 867152707 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19256 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19256
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18178
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>;tag=as7b7a0bfe
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 986277559 986277559 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18178 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>;tag=as7b7a0bfe
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>;tag=as7b7a0bfe
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK06919d78;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2b00650f
To: <sip:0638708675@172.22.11.181>;tag=as7b7a0bfe
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0c21a14a4fe09b465dbcba0d3c0f2467@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:36:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 926150395 926150395 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17442 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 26bb368347ea9123429199714a510617@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17442
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11886
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>;tag=as661d9091
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 874888659 874888659 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11886 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '205be52e1423944365b48e915c438f48@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '26bb368347ea9123429199714a510617@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>;tag=as661d9091
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72639f9b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as689ed67e
To: <sip:0638708675@172.22.11.181>;tag=as661d9091
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 26bb368347ea9123429199714a510617@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1ba1adb6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:36:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1742882032 1742882032 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10610 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10610
Looking for 0935819802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1ba1adb6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1ba1adb6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19766
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1ba1adb6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>;tag=as3e9c4bad
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 141208436 141208436 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19766 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19766
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1ba1adb6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>;tag=as3e9c4bad
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 141208436 141208437 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19766 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6597948b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>;tag=as3e9c4bad
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '26bb368347ea9123429199714a510617@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5f20f030;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>;tag=as3e9c4bad
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5f20f030;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0086536e
To: <sip:0935819802@172.22.11.181>;tag=as3e9c4bad
Call-ID: 67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c491126;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:38:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1622838733 1622838733 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15218 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15218
Looking for 0935819802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c491126;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c491126;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c491126;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>;tag=as3c6a3a03
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1030921520 1030921520 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19060 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c491126;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>;tag=as3c6a3a03
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1030921520 1030921521 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19060 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK63a82c01;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
To: <sip:0935819802@172.22.11.181>;tag=as3c6a3a03
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '67e3dbd50484c4ef69c104834c5b9b67@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:38:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1079550149 1079550149 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14254 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14254
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15418
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>;tag=as1a80f038
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1769097646 1769097646 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15418 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '528b956429d9d38815c27d44641122d4@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK46fa70eb;rport
Max-Forwards: 70
From: <sip:0935819802@172.22.11.181>;tag=as3c6a3a03
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK46fa70eb;received=172.22.11.181;rport=5060
From: <sip:0935819802@172.22.11.181>;tag=as3c6a3a03
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as054c59c4
Call-ID: 528b956429d9d38815c27d44641122d4@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '528b956429d9d38815c27d44641122d4@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>;tag=as1a80f038
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>;tag=as1a80f038
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK392fa936;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as671d600a
To: <sip:0638708675@172.22.11.181>;tag=as1a80f038
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4487dbbd3a243b260f82c1ce69a5184e@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:39:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1767562779 1767562779 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10614 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10614
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10584
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>;tag=as07117bb5
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2104410648 2104410648 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10584 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>;tag=as07117bb5
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>;tag=as07117bb5
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK505309f0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46f2c473
To: <sip:0638708675@172.22.11.181>;tag=as07117bb5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '10d5b5167704466d11a39b3722ec5ea8@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:39:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 159138832 159138832 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15916 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15916
Looking for 0637990999 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18440
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>;tag=as449f6337
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1316358818 1316358818 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18440 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>;tag=as449f6337
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>;tag=as449f6337
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637990999@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK76ed621a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60f36dc0
To: <sip:0637990999@172.22.11.181>;tag=as449f6337
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 41dccefc207251512c4f89af542e95ee@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '41dccefc207251512c4f89af542e95ee@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK30f33c41;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:42:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2091076077 2091076077 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11658 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11658
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK30f33c41;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK30f33c41;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11088
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK30f33c41;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>;tag=as6dfe7da9
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1595543290 1595543290 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11088 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11088
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK30f33c41;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>;tag=as6dfe7da9
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1595543290 1595543291 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11088 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK457902db;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
To: <sip:0939011286@172.22.11.181>;tag=as6dfe7da9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK16366427;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as6dfe7da9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK16366427;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as6dfe7da9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as51a547bf
Call-ID: 139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '139cf60266a70cdf70b3b06327979cd3@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK11b3caf0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:46:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1057752389 1057752389 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10524 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10524
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK11b3caf0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK11b3caf0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16466
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK11b3caf0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>;tag=as50b5c7f9
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2043864217 2043864217 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16466 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16466
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK11b3caf0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>;tag=as50b5c7f9
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2043864217 2043864218 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16466 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK54a211c4;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
To: <sip:0939011286@172.22.11.181>;tag=as50b5c7f9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK62972cb9;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as50b5c7f9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK62972cb9;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as50b5c7f9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e17015f
Call-ID: 48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '48413dea3face7db2eb2fd9922722cad@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0933622474@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6c56b96f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:50:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 620571792 620571792 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13420 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13420
Looking for 0933622474 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6c56b96f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6c56b96f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18870
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6c56b96f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>;tag=as3e3b09ad
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1216740190 1216740190 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18870 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18870
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6c56b96f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>;tag=as3e3b09ad
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1216740190 1216740191 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18870 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0933622474@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK397a1b2e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
To: <sip:0933622474@172.22.11.181>;tag=as3e3b09ad
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK113b6f9f;rport
Max-Forwards: 70
From: <sip:0933622474@172.22.11.181>;tag=as3e3b09ad
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK113b6f9f;received=172.22.11.181;rport=5060
From: <sip:0933622474@172.22.11.181>;tag=as3e3b09ad
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1966753e
Call-ID: 61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '61445e4907b1cb5e67ff383923fea6ad@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0933622474@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK65235bf6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:54:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 238710352 238710352 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15834 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15834
Looking for 0933622474 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK65235bf6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK65235bf6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18914
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK65235bf6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>;tag=as7196fe84
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 993782909 993782909 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18914 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18914
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK65235bf6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>;tag=as7196fe84
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0933622474@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 993782909 993782910 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18914 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0933622474@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c4deee8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
To: <sip:0933622474@172.22.11.181>;tag=as7196fe84
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK073687eb;rport
Max-Forwards: 70
From: <sip:0933622474@172.22.11.181>;tag=as7196fe84
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK073687eb;received=172.22.11.181;rport=5060
From: <sip:0933622474@172.22.11.181>;tag=as7196fe84
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as022e7f36
Call-ID: 473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '473d38e4667c0bb70e2d1c9a61e3c732@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:56:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 192204020 192204020 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11518 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11518
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13792
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>;tag=as2ed727e1
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 652707799 652707799 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13792 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>;tag=as2ed727e1
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>;tag=as2ed727e1
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK177294b3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6e54643d
To: <sip:0939011286@172.22.11.181>;tag=as2ed727e1
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '47b6ab716e4aafc00348dcbd3fdef45a@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636602640@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:56:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1210655580 1210655580 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15950 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15950
Looking for 0636602640 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>;tag=as0815f3d9
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 892461828 892461828 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14938 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0636602640@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>;tag=as0815f3d9
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>;tag=as0815f3d9
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636602640@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK15d028f6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7c4d4225
To: <sip:0636602640@172.22.11.181>;tag=as0815f3d9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '41d7428e530cf58519dd0aff71e2bcec@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK39568ce2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:56:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 350804073 350804073 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12158 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12158
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK39568ce2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK39568ce2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10518
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK39568ce2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>;tag=as4c121946
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 617625170 617625170 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10518 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10518
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK39568ce2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>;tag=as4c121946
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 617625170 617625171 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10518 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4cfa3aeb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
To: <sip:0939011286@172.22.11.181>;tag=as4c121946
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636602640@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50bf9b1a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:57:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1043274265 1043274265 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19372 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19372
Looking for 0636602640 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50bf9b1a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50bf9b1a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12306
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50bf9b1a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>;tag=as28ce4df9
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1940415998 1940415998 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12306 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '0751101c7dd06aaa392575171c722697@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52e782db;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as4c121946
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52e782db;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as4c121946
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7be8581e
Call-ID: 0751101c7dd06aaa392575171c722697@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '0751101c7dd06aaa392575171c722697@172.22.11.142:5060' Method: ACK
Audio is at 12306
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50bf9b1a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>;tag=as28ce4df9
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1940415998 1940415999 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12306 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636602640@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK05d31971;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
To: <sip:0636602640@172.22.11.181>;tag=as28ce4df9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:58:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1757105622 1757105622 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17782 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17782
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16012
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>;tag=as5521b9bc
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 286741036 286741036 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16012 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>;tag=as5521b9bc
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>;tag=as5521b9bc
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK605dfcd3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as114674c1
To: <sip:0939011286@172.22.11.181>;tag=as5521b9bc
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '75d0c2d3118fb6ca7a7516371899ab49@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5732cb6d;rport
Max-Forwards: 70
From: <sip:0636602640@172.22.11.181>;tag=as28ce4df9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5732cb6d;received=172.22.11.181;rport=5060
From: <sip:0636602640@172.22.11.181>;tag=as28ce4df9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7fba08e2
Call-ID: 1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1fba8f916a2c5e557c1d14105b4cb692@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK49644921;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 06:59:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 501868193 501868193 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12582 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12582
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK49644921;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK49644921;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK49644921;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>;tag=as4e8c775c
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1730284910 1730284910 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12610 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK49644921;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>;tag=as4e8c775c
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1730284910 1730284911 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12610 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0ebe4054;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
To: <sip:0939011286@172.22.11.181>;tag=as4e8c775c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '77ceadb403992f63485fd3836494472b@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK69f80bb7;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as4e8c775c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK69f80bb7;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as4e8c775c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as44405097
Call-ID: 77ceadb403992f63485fd3836494472b@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '77ceadb403992f63485fd3836494472b@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e42a4ae;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:04:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 811796167 811796167 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11700 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11700
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e42a4ae;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e42a4ae;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13158
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e42a4ae;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>;tag=as289eb534
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1861108587 1861108587 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13158 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13158
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e42a4ae;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>;tag=as289eb534
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1861108587 1861108588 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13158 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5801b5fd;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
To: <sip:0939011286@172.22.11.181>;tag=as289eb534
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1c83f719;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as289eb534
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1c83f719;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as289eb534
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0e9217fe
Call-ID: 2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2a47f7d2703373b725a100d2086360e5@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cf19b36;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:08:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 504011473 504011473 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18344 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18344
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cf19b36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cf19b36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19712
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cf19b36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>;tag=as27d8f18f
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1253277955 1253277955 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19712 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19712
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cf19b36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>;tag=as27d8f18f
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1253277955 1253277956 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19712 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK70644528;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
To: <sip:0939011286@172.22.11.181>;tag=as27d8f18f
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '48ecce4d14f66312614a76f15accb942@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK49e537f1;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as27d8f18f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK49e537f1;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as27d8f18f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2c22a948
Call-ID: 48ecce4d14f66312614a76f15accb942@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '48ecce4d14f66312614a76f15accb942@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK20bdabcc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:25:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 683649661 683649661 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14968 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 468432a302957672264456ed691ec05b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14968
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK20bdabcc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK20bdabcc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10496
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK20bdabcc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>;tag=as248e3a34
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1403474581 1403474581 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10496 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10496
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK20bdabcc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>;tag=as248e3a34
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1403474581 1403474582 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10496 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2c615f6d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>;tag=as248e3a34
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5172d576;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>;tag=as248e3a34
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '468432a302957672264456ed691ec05b@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5172d576;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d683044
To: <sip:0939319680@172.22.11.181>;tag=as248e3a34
Call-ID: 468432a302957672264456ed691ec05b@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '468432a302957672264456ed691ec05b@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634758868@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3c031668;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:33:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2028110766 2028110766 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14746 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14746
Looking for 0634758868 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3c031668;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634758868@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3c031668;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634758868@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11590
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3c031668;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>;tag=as6654a651
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634758868@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 786434924 786434924 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11590 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11590
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3c031668;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>;tag=as6654a651
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634758868@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 786434924 786434925 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11590 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634758868@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5cc72c3f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
To: <sip:0634758868@172.22.11.181>;tag=as6654a651
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK348b8533;rport
Max-Forwards: 70
From: <sip:0634758868@172.22.11.181>;tag=as6654a651
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK348b8533;received=172.22.11.181;rport=5060
From: <sip:0634758868@172.22.11.181>;tag=as6654a651
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as045a9fc5
Call-ID: 30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '30f091bf1284a68f3358fba80b1ae105@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK371d3a45;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:34:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1256739524 1256739524 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15820 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15820
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK371d3a45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK371d3a45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK371d3a45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>;tag=as25014b91
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1897096407 1897096407 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15002 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK371d3a45;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>;tag=as25014b91
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1897096407 1897096408 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15002 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7dca08a3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
To: <sip:0939319680@172.22.11.181>;tag=as25014b91
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '6b929ead4660331141d075e2636b2b78@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK79c47960;rport
Max-Forwards: 70
From: <sip:0939319680@172.22.11.181>;tag=as25014b91
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK79c47960;received=172.22.11.181;rport=5060
From: <sip:0939319680@172.22.11.181>;tag=as25014b91
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0f743d3e
Call-ID: 6b929ead4660331141d075e2636b2b78@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '6b929ead4660331141d075e2636b2b78@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0638708675@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK42b79b8b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 07:36:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 906612833 906612833 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11944 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11944
Looking for 0638708675 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK42b79b8b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK42b79b8b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11826
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK42b79b8b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>;tag=as342acf6c
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 668153194 668153194 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11826 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11826
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK42b79b8b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>;tag=as342acf6c
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0638708675@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 668153194 668153195 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11826 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2fd3908e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>;tag=as342acf6c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0638708675@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6bbc80da;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>;tag=as342acf6c
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6bbc80da;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19c658be
To: <sip:0638708675@172.22.11.181>;tag=as342acf6c
Call-ID: 47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '47d3106d3d02f55a181bfa3b41dbc1a2@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:00:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2017450536 2017450536 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13554 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13554
Looking for 0935819802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18258
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>;tag=as146012ef
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1770114219 1770114219 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18258 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>;tag=as146012ef
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>;tag=as146012ef
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3bebc821;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3bab0659
To: <sip:0935819802@172.22.11.181>;tag=as146012ef
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 75e18f072190b50012bb1797255785e7@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '75e18f072190b50012bb1797255785e7@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77dcaaf7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:02:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1786231064 1786231064 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13536 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13536
Looking for 0935819802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77dcaaf7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77dcaaf7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12208
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77dcaaf7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>;tag=as4d1b73f3
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1091423749 1091423749 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12208
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77dcaaf7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>;tag=as4d1b73f3
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1091423749 1091423750 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12208 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4ca4589a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>;tag=as4d1b73f3
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931661267@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5c52e9e5;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:03:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 949594200 949594200 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10208 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10208
Looking for 0931661267 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5c52e9e5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5c52e9e5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14748
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5c52e9e5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>;tag=as7d38cef5
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1849621862 1849621862 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14748 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14748
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5c52e9e5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>;tag=as7d38cef5
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931661267@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1849621862 1849621863 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14748 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931661267@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK75bc923a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
To: <sip:0931661267@172.22.11.181>;tag=as7d38cef5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7445a3f9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>;tag=as4d1b73f3
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7445a3f9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fabcb2f
To: <sip:0935819802@172.22.11.181>;tag=as4d1b73f3
Call-ID: 32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935819802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4616d3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:03:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 781545356 781545356 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13898 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13898
Looking for 0935819802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4616d3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4616d3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11786
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4616d3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>;tag=as3bcd9a0a
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 448966866 448966866 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11786 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK14955b94;rport
Max-Forwards: 70
From: <sip:0931661267@172.22.11.181>;tag=as7d38cef5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK14955b94;received=172.22.11.181;rport=5060
From: <sip:0931661267@172.22.11.181>;tag=as7d38cef5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10167672
Call-ID: 6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '6396c2471b9f55352f02a2570a23dc6e@172.22.11.142:5060' Method: ACK
Audio is at 11786
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4616d3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>;tag=as3bcd9a0a
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935819802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 448966866 448966867 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11786 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935819802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK343546cb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
To: <sip:0935819802@172.22.11.181>;tag=as3bcd9a0a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '076588282a222a0679a921bc011d6fa2@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2d741998;rport
Max-Forwards: 70
From: <sip:0935819802@172.22.11.181>;tag=as3bcd9a0a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2d741998;received=172.22.11.181;rport=5060
From: <sip:0935819802@172.22.11.181>;tag=as3bcd9a0a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as052753d9
Call-ID: 076588282a222a0679a921bc011d6fa2@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '076588282a222a0679a921bc011d6fa2@172.22.11.142:5060' Method: ACK
Really destroying SIP dialog '32e1e4770dfbe0ac3ba935586e7d6dd2@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0630448249@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4e516eef;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:04:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1615776782 1615776782 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13700 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13700
Looking for 0630448249 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4e516eef;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4e516eef;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12218
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4e516eef;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>;tag=as790ce784
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 711219650 711219650 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12218 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12218
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4e516eef;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>;tag=as790ce784
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 711219650 711219651 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12218 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0630448249@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2c2ddcce;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
To: <sip:0630448249@172.22.11.181>;tag=as790ce784
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1e7c7000;rport
Max-Forwards: 70
From: <sip:0630448249@172.22.11.181>;tag=as790ce784
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1e7c7000;received=172.22.11.181;rport=5060
From: <sip:0630448249@172.22.11.181>;tag=as790ce784
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as03626b9f
Call-ID: 3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3af37cb7474f1a7777882fc542381a6e@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939319680@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK629e7bc0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:08:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 607561746 607561746 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11160 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11160
Looking for 0939319680 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK629e7bc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK629e7bc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12552
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK629e7bc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>;tag=as651bd4bc
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1612210691 1612210691 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12552 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12552
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK629e7bc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>;tag=as651bd4bc
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939319680@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1612210691 1612210692 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12552 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939319680@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK080e60f5;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
To: <sip:0939319680@172.22.11.181>;tag=as651bd4bc
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d31165c;rport
Max-Forwards: 70
From: <sip:0939319680@172.22.11.181>;tag=as651bd4bc
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d31165c;received=172.22.11.181;rport=5060
From: <sip:0939319680@172.22.11.181>;tag=as651bd4bc
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as19449073
Call-ID: 620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '620963ce51b898473aa88f4a1c8f8d4b@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636160153@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK133756cc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:09:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 855872963 855872963 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16984 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16984
Looking for 0636160153 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK133756cc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636160153@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK133756cc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636160153@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10822
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK133756cc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>;tag=as586c1c0a
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636160153@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1278771428 1278771428 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10822 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10822
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK133756cc;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>;tag=as586c1c0a
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636160153@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1278771428 1278771429 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10822 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636160153@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK423624a8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
To: <sip:0636160153@172.22.11.181>;tag=as586c1c0a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK0474dfb5;rport
Max-Forwards: 70
From: <sip:0636160153@172.22.11.181>;tag=as586c1c0a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK0474dfb5;received=172.22.11.181;rport=5060
From: <sip:0636160153@172.22.11.181>;tag=as586c1c0a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6918a580
Call-ID: 7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '7ac84a8e69ec947779dae1856e5e23b4@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935629226@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:24:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1612929972 1612929972 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18574 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18574
Looking for 0935629226 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935629226@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935629226@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17248
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>;tag=as26edc560
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935629226@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 843728595 843728595 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17248 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0935629226@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>;tag=as26edc560
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>;tag=as26edc560
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935629226@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55051b1d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10eb210d
To: <sip:0935629226@172.22.11.181>;tag=as26edc560
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1265d46a19c31c1a76c116a43ffaa43c@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0934770149@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10548966;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:44:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 185752661 185752661 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10926 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 34784b2267d69592258c81916305add4@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10926
Looking for 0934770149 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10548966;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934770149@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10548966;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934770149@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16468
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10548966;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>;tag=as07a020f8
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934770149@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1525572970 1525572970 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16468 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16468
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10548966;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>;tag=as07a020f8
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934770149@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1525572970 1525572971 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16468 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0934770149@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK258712fc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
To: <sip:0934770149@172.22.11.181>;tag=as07a020f8
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '34784b2267d69592258c81916305add4@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK093b6e65;rport
Max-Forwards: 70
From: <sip:0934770149@172.22.11.181>;tag=as07a020f8
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK093b6e65;received=172.22.11.181;rport=5060
From: <sip:0934770149@172.22.11.181>;tag=as07a020f8
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1f9b1aa8
Call-ID: 34784b2267d69592258c81916305add4@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '34784b2267d69592258c81916305add4@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0630448249@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27c95562;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:48:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 276696301 276696301 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18840 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18840
Looking for 0630448249 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27c95562;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27c95562;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15740
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27c95562;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>;tag=as41c9b45c
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 249899087 249899087 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15740 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15740
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27c95562;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>;tag=as41c9b45c
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 249899087 249899088 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15740 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0630448249@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7be8deec;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
To: <sip:0630448249@172.22.11.181>;tag=as41c9b45c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '010ab21108f9767019e044fc4b436709@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1722b351;rport
Max-Forwards: 70
From: <sip:0630448249@172.22.11.181>;tag=as41c9b45c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1722b351;received=172.22.11.181;rport=5060
From: <sip:0630448249@172.22.11.181>;tag=as41c9b45c
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as78305c6e
Call-ID: 010ab21108f9767019e044fc4b436709@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '010ab21108f9767019e044fc4b436709@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0633720802@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2ce0f802;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 08:55:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 164776309 164776309 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16230 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16230
Looking for 0633720802 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2ce0f802;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633720802@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2ce0f802;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633720802@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 17726
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2ce0f802;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>;tag=as71758c74
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633720802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1605667290 1605667290 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17726 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 17726
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2ce0f802;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>;tag=as71758c74
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633720802@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1605667290 1605667291 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 17726 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0633720802@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK475128dc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
To: <sip:0633720802@172.22.11.181>;tag=as71758c74
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '771257d64d3fd392234399711b9956cb@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK62ec7e02;rport
Max-Forwards: 70
From: <sip:0633720802@172.22.11.181>;tag=as71758c74
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK62ec7e02;received=172.22.11.181;rport=5060
From: <sip:0633720802@172.22.11.181>;tag=as71758c74
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4370df43
Call-ID: 771257d64d3fd392234399711b9956cb@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '771257d64d3fd392234399711b9956cb@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0630448249@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4269c99e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 09:05:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1539820434 1539820434 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13580 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13580
Looking for 0630448249 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4269c99e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4269c99e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16844
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4269c99e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>;tag=as49ec01bf
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1694179508 1694179508 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16844 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16844
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4269c99e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>;tag=as49ec01bf
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0630448249@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1694179508 1694179509 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16844 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0630448249@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2eda386b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>;tag=as49ec01bf
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0630448249@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f96a582;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>;tag=as49ec01bf
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '37308e6b386be8861f27fb626c815609@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f96a582;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1796af6a
To: <sip:0630448249@172.22.11.181>;tag=as49ec01bf
Call-ID: 37308e6b386be8861f27fb626c815609@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '37308e6b386be8861f27fb626c815609@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634306812@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0badbe59;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 10:00:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1716501649 1716501649 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16936 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16936
Looking for 0634306812 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0badbe59;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634306812@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0badbe59;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634306812@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0badbe59;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>;tag=as539badf1
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634306812@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1286116725 1286116725 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0badbe59;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>;tag=as539badf1
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634306812@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1286116725 1286116726 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19318 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634306812@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK55009827;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>;tag=as539badf1
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0634306812@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0c8da867;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>;tag=as539badf1
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0c8da867;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4b8e84ca
To: <sip:0634306812@172.22.11.181>;tag=as539badf1
Call-ID: 1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '1d6d9a6522abc5b1100367666ab6e4aa@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636602640@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23404c97;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 10:17:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1081252249 1081252249 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17272 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4e304a9334840da62db8597339826b87@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17272
Looking for 0636602640 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23404c97;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23404c97;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16278
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23404c97;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>;tag=as54a43a27
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 205777841 205777841 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16278 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16278
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK23404c97;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>;tag=as54a43a27
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636602640@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 205777841 205777842 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16278 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636602640@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3a62ce37;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>;tag=as54a43a27
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0636602640@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK47313f51;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>;tag=as54a43a27
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '4e304a9334840da62db8597339826b87@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK47313f51;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6b40be24
To: <sip:0636602640@172.22.11.181>;tag=as54a43a27
Call-ID: 4e304a9334840da62db8597339826b87@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '4e304a9334840da62db8597339826b87@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33efa75d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 10:39:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 299353923 299353923 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11150 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 66fc9d663033156262f590375193fd86@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11150
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33efa75d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33efa75d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19896
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33efa75d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>;tag=as51e046d8
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 640020909 640020909 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19896 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19896
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK33efa75d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>;tag=as51e046d8
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 640020909 640020910 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19896 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0be6fe4d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>;tag=as51e046d8
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7ffccde8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>;tag=as51e046d8
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '66fc9d663033156262f590375193fd86@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7ffccde8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as09e2bf3e
To: <sip:0939011286@172.22.11.181>;tag=as51e046d8
Call-ID: 66fc9d663033156262f590375193fd86@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '66fc9d663033156262f590375193fd86@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 10:55:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 427838130 427838130 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13006 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13006
Looking for 0637990999 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11652
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>;tag=as5fc7d61c
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 369312375 369312375 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11652 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>;tag=as5fc7d61c
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>;tag=as5fc7d61c
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637990999@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36834f5a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as056df39f
To: <sip:0637990999@172.22.11.181>;tag=as5fc7d61c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0b979a247ae570d244cf4dfc7cb8c53f@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939386966@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50ed90f1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:00:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1826163508 1826163508 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18924 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18924
Looking for 0939386966 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50ed90f1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939386966@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50ed90f1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939386966@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13244
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50ed90f1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>;tag=as44e02cd5
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939386966@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 407961550 407961550 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13244
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50ed90f1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>;tag=as44e02cd5
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939386966@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 407961550 407961551 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13244 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939386966@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6906b1e0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
To: <sip:0939386966@172.22.11.181>;tag=as44e02cd5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2b5ec01f;rport
Max-Forwards: 70
From: <sip:0939386966@172.22.11.181>;tag=as44e02cd5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2b5ec01f;received=172.22.11.181;rport=5060
From: <sip:0939386966@172.22.11.181>;tag=as44e02cd5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5d079233
Call-ID: 3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3c0c05ce27f0af0e1a1baf2b67d61993@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931043251@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK518906a2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:01:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 874664377 874664377 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14698 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14698
Looking for 0931043251 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK518906a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK518906a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19276
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK518906a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>;tag=as630bc26b
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 807671030 807671030 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19276 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19276
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK518906a2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>;tag=as630bc26b
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 807671030 807671031 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19276 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931043251@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK37e92f2f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
To: <sip:0931043251@172.22.11.181>;tag=as630bc26b
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK31ee2fcb;rport
Max-Forwards: 70
From: <sip:0931043251@172.22.11.181>;tag=as630bc26b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK31ee2fcb;received=172.22.11.181;rport=5060
From: <sip:0931043251@172.22.11.181>;tag=as630bc26b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as558b2a02
Call-ID: 2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2d9a790d6cf9f1615583d10874ec7f5b@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931043251@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1d85027d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:05:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1609922564 1609922564 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11502 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 781c256836eab046577b0ef736992733@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11502
Looking for 0931043251 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1d85027d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1d85027d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16096
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1d85027d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>;tag=as423adbe5
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1192792736 1192792736 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16096 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16096
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1d85027d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>;tag=as423adbe5
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1192792736 1192792737 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16096 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931043251@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK58b514e1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
To: <sip:0931043251@172.22.11.181>;tag=as423adbe5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '781c256836eab046577b0ef736992733@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47f71538;rport
Max-Forwards: 70
From: <sip:0931043251@172.22.11.181>;tag=as423adbe5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47f71538;received=172.22.11.181;rport=5060
From: <sip:0931043251@172.22.11.181>;tag=as423adbe5
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46695dc8
Call-ID: 781c256836eab046577b0ef736992733@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '781c256836eab046577b0ef736992733@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0930200653@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:11:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 777083622 777083622 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19574 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19574
Looking for 0930200653 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930200653@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930200653@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12690
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>;tag=as2b906727
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930200653@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 114938879 114938879 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12690 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>;tag=as2b906727
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0930200653@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bdeea44;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as70ca8e71
To: <sip:0930200653@172.22.11.181>;tag=as2b906727
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '10aee63172f381d254fb2e1763d9431b@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636223145@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:22:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1349842565 1349842565 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18550 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18550
Looking for 0636223145 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19802
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>;tag=as1c9f1808
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1854505901 1854505901 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19802 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0636223145@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>;tag=as1c9f1808
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>;tag=as1c9f1808
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636223145@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1f080b54;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3f9f3090
To: <sip:0636223145@172.22.11.181>;tag=as1c9f1808
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '256ca7f8235614d6708d5d9d584e8fdb@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635639751@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0a52d438;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:24:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1456080669 1456080669 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12614 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12614
Looking for 0635639751 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0a52d438;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635639751@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0a52d438;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635639751@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13222
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0a52d438;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>;tag=as7e1f89c2
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635639751@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 659978966 659978966 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13222 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0633200405@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6918f61f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:25:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1136578163 1136578163 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13976 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13976
Looking for 0633200405 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6918f61f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633200405@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6918f61f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633200405@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10052
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6918f61f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>;tag=as1d0d3205
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633200405@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 888298697 888298697 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10052 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13222
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0a52d438;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>;tag=as7e1f89c2
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635639751@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 659978966 659978967 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13222 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635639751@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK201b0f40;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>;tag=as7e1f89c2
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0635639751@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10540d40;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>;tag=as7e1f89c2
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10540d40;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5cf12dae
To: <sip:0635639751@172.22.11.181>;tag=as7e1f89c2
Call-ID: 4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '4e7c80d221f59b15021dcc2d0025520b@172.22.11.142:5060' Method: BYE
Audio is at 10052
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6918f61f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>;tag=as1d0d3205
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0633200405@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 888298697 888298698 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10052 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0633200405@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK25a0aaa4;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>;tag=as1d0d3205
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0633200405@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK78f8345d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>;tag=as1d0d3205
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '5754277523d47e8246af6b811f7082b6@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK78f8345d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0d1b41f2
To: <sip:0633200405@172.22.11.181>;tag=as1d0d3205
Call-ID: 5754277523d47e8246af6b811f7082b6@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '5754277523d47e8246af6b811f7082b6@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636223145@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0eb649b7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:35:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1941446992 1941446992 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14898 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14898
Looking for 0636223145 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0eb649b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0eb649b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13248
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0eb649b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>;tag=as2ae31f4f
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1056123698 1056123698 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13248 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13248
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0eb649b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>;tag=as2ae31f4f
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636223145@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1056123698 1056123699 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13248 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636223145@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3475fdbc;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>;tag=as2ae31f4f
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0636223145@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0c17d7d5;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>;tag=as2ae31f4f
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0c17d7d5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as71f9bd8c
To: <sip:0636223145@172.22.11.181>;tag=as2ae31f4f
Call-ID: 1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '1fa0025e5a8870aa39bf6e6262fcc039@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:42:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 223803917 223803917 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12724 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12724
Looking for 0637588264 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16508
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>;tag=as520bb3a6
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1877227456 1877227456 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16508 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>;tag=as520bb3a6
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>;tag=as520bb3a6
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637588264@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK32495ff2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56c21f3e
To: <sip:0637588264@172.22.11.181>;tag=as520bb3a6
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2f9cffd347bcec9f74221fd73fabebea@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0938047792@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4594fd;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:44:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1404338357 1404338357 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17818 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17818
Looking for 0938047792 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4594fd;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4594fd;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12946
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4594fd;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>;tag=as4ba43a7e
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 943289921 943289921 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12946 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12946
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3f4594fd;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>;tag=as4ba43a7e
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 943289921 943289922 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12946 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0938047792@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK038972c0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
To: <sip:0938047792@172.22.11.181>;tag=as4ba43a7e
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:45:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1800502993 1800502993 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15826 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15826
Looking for 0637588264 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15558
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>;tag=as2e7c9be7
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 440326803 440326803 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15558 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>;tag=as2e7c9be7
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>;tag=as2e7c9be7
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637588264@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK04438431;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1bccf889
To: <sip:0637588264@172.22.11.181>;tag=as2e7c9be7
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2f0b64d2175fdc99623a33e313e3ad43@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '6d289958213a725a253d5407195b4ed4@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK07cdcf31;rport
Max-Forwards: 70
From: <sip:0938047792@172.22.11.181>;tag=as4ba43a7e
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK07cdcf31;received=172.22.11.181;rport=5060
From: <sip:0938047792@172.22.11.181>;tag=as4ba43a7e
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2263a580
Call-ID: 6d289958213a725a253d5407195b4ed4@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '6d289958213a725a253d5407195b4ed4@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637990999@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77384adf;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:47:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1082154009 1082154009 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14876 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14876
Looking for 0637990999 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77384adf;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77384adf;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15012
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77384adf;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>;tag=as3219e219
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1384893228 1384893228 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15012 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15012
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK77384adf;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>;tag=as3219e219
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637990999@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1384893228 1384893229 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15012 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637990999@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK21ef210f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>;tag=as3219e219
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0637990999@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK01752ab3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>;tag=as3219e219
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK01752ab3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7e2f4f89
To: <sip:0637990999@172.22.11.181>;tag=as3219e219
Call-ID: 1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '1843e0a21fe6847b3bcbd3f7325ba42c@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634953259@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0f3bdb36;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:51:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 702569388 702569388 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17304 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17304
Looking for 0634953259 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0f3bdb36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634953259@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0f3bdb36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634953259@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19530
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0f3bdb36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>;tag=as2a20bc3a
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634953259@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 609634076 609634076 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19530 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19530
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0f3bdb36;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>;tag=as2a20bc3a
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634953259@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 609634076 609634077 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19530 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634953259@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3a8d75fa;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
To: <sip:0634953259@172.22.11.181>;tag=as2a20bc3a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK29c6625e;rport
Max-Forwards: 70
From: <sip:0634953259@172.22.11.181>;tag=as2a20bc3a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK29c6625e;received=172.22.11.181;rport=5060
From: <sip:0634953259@172.22.11.181>;tag=as2a20bc3a
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2a661946
Call-ID: 1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1d2e0c2a2200c451163e6ab80b05a81c@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0938047792@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bd74e52;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 11:53:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 84455886 84455886 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10624 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10624
Looking for 0938047792 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bd74e52;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bd74e52;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18072
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bd74e52;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>;tag=as32abdf59
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 463430512 463430512 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18072 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18072
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7bd74e52;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>;tag=as32abdf59
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 463430512 463430513 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18072 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0938047792@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5da29603;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
To: <sip:0938047792@172.22.11.181>;tag=as32abdf59
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '42b8fe044142775c286829b445aae73f@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK192e9501;rport
Max-Forwards: 70
From: <sip:0938047792@172.22.11.181>;tag=as32abdf59
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK192e9501;received=172.22.11.181;rport=5060
From: <sip:0938047792@172.22.11.181>;tag=as32abdf59
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c3b2287
Call-ID: 42b8fe044142775c286829b445aae73f@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '42b8fe044142775c286829b445aae73f@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:10:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 65848793 65848793 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10654 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10654
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13118
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>;tag=as4fff1e1c
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 502301074 502301074 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13118 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>;tag=as4fff1e1c
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6dcd99e3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as61483e9c
To: <sip:0635714975@172.22.11.181>;tag=as4fff1e1c
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b390fe9;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:10:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1203245490 1203245490 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 12544 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:12544
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b390fe9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b390fe9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14986
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b390fe9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>;tag=as5378f742
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 552807153 552807153 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14986 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14986
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b390fe9;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>;tag=as5378f742
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 552807153 552807154 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14986 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK07b9ecee;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
To: <sip:0635714975@172.22.11.181>;tag=as5378f742
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '342450ad0016f74d7f5d61af6019ebf3@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6cb680f7;rport
Max-Forwards: 70
From: <sip:0635714975@172.22.11.181>;tag=as5378f742
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6cb680f7;received=172.22.11.181;rport=5060
From: <sip:0635714975@172.22.11.181>;tag=as5378f742
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as76e1360a
Call-ID: 38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '38bffc9930ddc70f19e8f0ec63959542@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34eff869;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:19:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1826117374 1826117374 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15502 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15502
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34eff869;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34eff869;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12666
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34eff869;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>;tag=as5d222326
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2126596329 2126596329 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12666 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12666
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34eff869;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>;tag=as5d222326
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2126596329 2126596330 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12666 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK595450ec;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
To: <sip:0935288746@172.22.11.181>;tag=as5d222326
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2e7cd948;rport
Max-Forwards: 70
From: <sip:0935288746@172.22.11.181>;tag=as5d222326
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK2e7cd948;received=172.22.11.181;rport=5060
From: <sip:0935288746@172.22.11.181>;tag=as5d222326
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as110b5ec0
Call-ID: 1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1162b1485b1a47fd0e5c212b5e9f87fa@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0631497299@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:21:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1737309334 1737309334 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13386 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13386
Looking for 0631497299 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631497299@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631497299@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19504
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>;tag=as61c730ec
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0631497299@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1215713439 1215713439 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19504 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0631497299@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>;tag=as61c730ec
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>;tag=as61c730ec
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0631497299@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b8d849;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2bde8f73
To: <sip:0631497299@172.22.11.181>;tag=as61c730ec
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2323780412dcf9973dd04c421e660847@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2323780412dcf9973dd04c421e660847@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634121440@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK59ff178b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:45:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2109179690 2109179690 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14558 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14558
Looking for 0634121440 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK59ff178b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK59ff178b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12202
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK59ff178b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>;tag=as720f2b3a
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1265668641 1265668641 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12202 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12202
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK59ff178b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>;tag=as720f2b3a
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1265668641 1265668642 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12202 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK38ec95c8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>;tag=as720f2b3a
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931209365@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36c9c6a4;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:46:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1974109549 1974109549 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19942 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19942
Looking for 0931209365 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36c9c6a4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36c9c6a4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18314
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36c9c6a4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>;tag=as6d6401c9
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 741352878 741352878 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18314 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK16ab6bc2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>;tag=as720f2b3a
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK16ab6bc2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0fcd4053
To: <sip:0634121440@172.22.11.181>;tag=as720f2b3a
Call-ID: 0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634121440@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d68cd23;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1791198081 1791198081 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15296 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 287469316c890eab384082355e778268@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15296
Looking for 0634121440 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d68cd23;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d68cd23;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15732
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d68cd23;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>;tag=as76498f75
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1845497992 1845497992 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15732 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18314
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK36c9c6a4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>;tag=as6d6401c9
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 741352878 741352879 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18314 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931209365@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6383d77b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
To: <sip:0931209365@172.22.11.181>;tag=as6d6401c9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Audio is at 15732
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d68cd23;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>;tag=as76498f75
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1845497992 1845497993 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15732 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK16aeb6c3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>;tag=as76498f75
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0b6fe3362e3a3f23724b16313bafabdd@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64d9b4fb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>;tag=as76498f75
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '287469316c890eab384082355e778268@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64d9b4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as564fabe6
To: <sip:0634121440@172.22.11.181>;tag=as76498f75
Call-ID: 287469316c890eab384082355e778268@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2afa26847035648935b7205a6d47f117@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK68abd6cb;rport
Max-Forwards: 70
From: <sip:0931209365@172.22.11.181>;tag=as6d6401c9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK68abd6cb;received=172.22.11.181;rport=5060
From: <sip:0931209365@172.22.11.181>;tag=as6d6401c9
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as437c49bf
Call-ID: 2afa26847035648935b7205a6d47f117@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2afa26847035648935b7205a6d47f117@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634121440@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e9bb526;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:48:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1873557210 1873557210 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18620 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18620
Looking for 0634121440 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e9bb526;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e9bb526;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12404
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e9bb526;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>;tag=as4f18de29
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 857191973 857191973 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12404 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12404
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6e9bb526;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>;tag=as4f18de29
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 857191973 857191974 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12404 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1c6a283b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
To: <sip:0634121440@172.22.11.181>;tag=as4f18de29
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '287469316c890eab384082355e778268@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:48:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2097956410 2097956410 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11006 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11006
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10600
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>;tag=as6f2b5ab5
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1324742692 1324742692 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10600 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d70a494;rport
Max-Forwards: 70
From: <sip:0634121440@172.22.11.181>;tag=as4f18de29
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d70a494;received=172.22.11.181;rport=5060
From: <sip:0634121440@172.22.11.181>;tag=as4f18de29
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as77db7590
Call-ID: 50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '50ab224b091184d666f7b8d017a4d896@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>;tag=as6f2b5ab5
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>;tag=as6f2b5ab5
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cfcb4fb;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50ada07d
To: <sip:0935288746@172.22.11.181>;tag=as6f2b5ab5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 51214fbe3543454f075022361e9b2785@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '51214fbe3543454f075022361e9b2785@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:49:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1062930052 1062930052 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11040 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11040
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19582
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>;tag=as37616841
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1293670237 1293670237 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19582 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>;tag=as37616841
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>;tag=as37616841
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2e00a6df;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as50099337
To: <sip:0935288746@172.22.11.181>;tag=as37616841
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6a0b7bda0d48fe9c0b58fee97db9df4c@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 12:55:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 419502894 419502894 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18142 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18142
Looking for 0637588264 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18568
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>;tag=as481b3cd5
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0637588264@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 697330677 697330677 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18568 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0637588264@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>;tag=as481b3cd5
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>;tag=as481b3cd5
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0637588264@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4c1bf81f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as00f691ba
To: <sip:0637588264@172.22.11.181>;tag=as481b3cd5
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0a79cde17b58d4941050dcdc20e8866d@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4656fc7e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:09:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1018442469 1018442469 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18204 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18204
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4656fc7e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4656fc7e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11598
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4656fc7e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>;tag=as027c79da
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 508470038 508470038 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11598 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11598
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4656fc7e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>;tag=as027c79da
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 508470038 508470039 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11598 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1a8a6b0a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
To: <sip:0635714975@172.22.11.181>;tag=as027c79da
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK35655060;rport
Max-Forwards: 70
From: <sip:0635714975@172.22.11.181>;tag=as027c79da
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK35655060;received=172.22.11.181;rport=5060
From: <sip:0635714975@172.22.11.181>;tag=as027c79da
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as480b760b
Call-ID: 32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '32fa8a3d722838774443921c317cf6eb@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64a46c34;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:22:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1084320546 1084320546 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16424 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 686c662373740afd532000292c85e57f@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16424
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64a46c34;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64a46c34;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 19322
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64a46c34;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>;tag=as65430d38
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 827598342 827598342 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19322 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 19322
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK64a46c34;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>;tag=as65430d38
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 827598342 827598343 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 19322 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK17fa8903;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>;tag=as65430d38
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0934853470@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:23:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1740051336 1740051336 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18748 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18748
Looking for 0934853470 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>;tag=as485f3d15
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 609620813 609620813 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11002 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK513a55ee;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>;tag=as65430d38
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '686c662373740afd532000292c85e57f@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK513a55ee;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as6d8bd1b6
To: <sip:0935183950@172.22.11.181>;tag=as65430d38
Call-ID: 686c662373740afd532000292c85e57f@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>;tag=as485f3d15
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0934853470@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5184e7d8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as58332ce0
To: <sip:0934853470@172.22.11.181>;tag=as485f3d15
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '686c662373740afd532000292c85e57f@172.22.11.142:5060' Method: BYE
Really destroying SIP dialog '5f07eadc7ba05e5076e3323470c1fe95@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f7c2f0c;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:28:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 191805408 191805408 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11488 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11488
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f7c2f0c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f7c2f0c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18028
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f7c2f0c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>;tag=as25334e38
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1542313905 1542313905 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18028 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18028
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f7c2f0c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>;tag=as25334e38
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1542313905 1542313906 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18028 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK43167e84;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>;tag=as25334e38
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK363599d3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>;tag=as25334e38
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '7f216eab6983641311cae26845dda7da@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK363599d3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3d43a0c3
To: <sip:0935183950@172.22.11.181>;tag=as25334e38
Call-ID: 7f216eab6983641311cae26845dda7da@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '7f216eab6983641311cae26845dda7da@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK658e3ee2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:31:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 228268247 228268247 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18006 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18006
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK658e3ee2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK658e3ee2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK658e3ee2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>;tag=as1950f9df
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2031023608 2031023608 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18610 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18610
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK658e3ee2;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>;tag=as1950f9df
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2031023608 2031023609 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18610 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK46c7eca0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
To: <sip:0935183950@172.22.11.181>;tag=as1950f9df
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5199b522;rport
Max-Forwards: 70
From: <sip:0935183950@172.22.11.181>;tag=as1950f9df
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK5199b522;received=172.22.11.181;rport=5060
From: <sip:0935183950@172.22.11.181>;tag=as1950f9df
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as60ea3676
Call-ID: 5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '5a19112c6def7e8b14ec08bf7603f043@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK334d8ca4;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:40:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1556402780 1556402780 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11128 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 37067977419c71520527293e57d675e1@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11128
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK334d8ca4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK334d8ca4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11000
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK334d8ca4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>;tag=as5ccae9f0
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 817249928 817249928 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11000
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK334d8ca4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>;tag=as5ccae9f0
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 817249928 817249929 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11000 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0509bdba;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>;tag=as5ccae9f0
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK53e3d7ab;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>;tag=as5ccae9f0
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '37067977419c71520527293e57d675e1@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK53e3d7ab;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as15c87454
To: <sip:0935183950@172.22.11.181>;tag=as5ccae9f0
Call-ID: 37067977419c71520527293e57d675e1@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '37067977419c71520527293e57d675e1@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931043251@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK335f345b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 13:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 13544889 13544889 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14186 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14186
Looking for 0931043251 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK335f345b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK335f345b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16098
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK335f345b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>;tag=as2b51d81b
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 672791407 672791407 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16098 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16098
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK335f345b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>;tag=as2b51d81b
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 672791407 672791408 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16098 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931043251@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK53c53a5a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
To: <sip:0931043251@172.22.11.181>;tag=as2b51d81b
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1e0288cc;rport
Max-Forwards: 70
From: <sip:0931043251@172.22.11.181>;tag=as2b51d81b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK1e0288cc;received=172.22.11.181;rport=5060
From: <sip:0931043251@172.22.11.181>;tag=as2b51d81b
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as2ad5d687
Call-ID: 785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '785fb728115ff4b603d15be468c82e9d@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0634121440@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b34311;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 14:01:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 393097860 393097860 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16858 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16858
Looking for 0634121440 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b34311;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b34311;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13514
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b34311;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>;tag=as4e366654
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1738369615 1738369615 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13514 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13514
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK34b34311;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>;tag=as4e366654
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0634121440@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1738369615 1738369616 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13514 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0634121440@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1516efc7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
To: <sip:0634121440@172.22.11.181>;tag=as4e366654
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK4026ab6c;rport
Max-Forwards: 70
From: <sip:0634121440@172.22.11.181>;tag=as4e366654
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK4026ab6c;received=172.22.11.181;rport=5060
From: <sip:0634121440@172.22.11.181>;tag=as4e366654
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as79d499be
Call-ID: 3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3253ccf134acb6da03fd221a67ee74a7@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0934853470@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b8c19f3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 14:47:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 44623057 44623057 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 11732 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:11732
Looking for 0934853470 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b8c19f3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b8c19f3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13922
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b8c19f3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>;tag=as1ad31f02
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 394226059 394226059 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13922 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13922
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b8c19f3;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>;tag=as1ad31f02
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0934853470@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 394226059 394226060 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13922 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0934853470@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK27e2a6ff;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
To: <sip:0934853470@172.22.11.181>;tag=as1ad31f02
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK553d0668;rport
Max-Forwards: 70
From: <sip:0934853470@172.22.11.181>;tag=as1ad31f02
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK553d0668;received=172.22.11.181;rport=5060
From: <sip:0934853470@172.22.11.181>;tag=as1ad31f02
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as024f2ed8
Call-ID: 4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '4c1b8da049a97be171dc5498429c651e@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931209365@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4dbaf199;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 15:23:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 654270486 654270486 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19520 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19520
Looking for 0931209365 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4dbaf199;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4dbaf199;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18924
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4dbaf199;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>;tag=as6c95ebeb
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1902800040 1902800040 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18924 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18924
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4dbaf199;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>;tag=as6c95ebeb
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931209365@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1902800040 1902800041 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18924 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931209365@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6b322289;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
To: <sip:0931209365@172.22.11.181>;tag=as6c95ebeb
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK7ad31346;rport
Max-Forwards: 70
From: <sip:0931209365@172.22.11.181>;tag=as6c95ebeb
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK7ad31346;received=172.22.11.181;rport=5060
From: <sip:0931209365@172.22.11.181>;tag=as6c95ebeb
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as10a1191b
Call-ID: 023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '023001fb61d58e7045fb906f5cf2d964@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0938047792@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50f1990c;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 15:37:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1376239500 1376239500 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10014 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 25c34c5a484621671b675800255b811b@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10014
Looking for 0938047792 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50f1990c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50f1990c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15972
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50f1990c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>;tag=as7ac6df66
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 788141024 788141024 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15972 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15972
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK50f1990c;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>;tag=as7ac6df66
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 788141024 788141025 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15972 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0938047792@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3a31f13d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
To: <sip:0938047792@172.22.11.181>;tag=as7ac6df66
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '25c34c5a484621671b675800255b811b@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK18c5d54c;rport
Max-Forwards: 70
From: <sip:0938047792@172.22.11.181>;tag=as7ac6df66
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK18c5d54c;received=172.22.11.181;rport=5060
From: <sip:0938047792@172.22.11.181>;tag=as7ac6df66
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1c4f8d34
Call-ID: 25c34c5a484621671b675800255b811b@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '25c34c5a484621671b675800255b811b@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0938047792@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4a5b906f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 15:47:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 820192932 820192932 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13984 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13984
Looking for 0938047792 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4a5b906f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4a5b906f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12642
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4a5b906f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>;tag=as484794ae
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1836770023 1836770023 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12642 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12642
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK4a5b906f;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>;tag=as484794ae
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938047792@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1836770023 1836770024 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12642 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0938047792@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6cda4d89;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
To: <sip:0938047792@172.22.11.181>;tag=as484794ae
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52c7dedb;rport
Max-Forwards: 70
From: <sip:0938047792@172.22.11.181>;tag=as484794ae
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52c7dedb;received=172.22.11.181;rport=5060
From: <sip:0938047792@172.22.11.181>;tag=as484794ae
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as411adf0d
Call-ID: 58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '58d281822b4b583f1d8204cc5ff45419@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0930235369@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 15:49:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1764526627 1764526627 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14974 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 07561b073b639599302ab7d716925671@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14974
Looking for 0930235369 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930235369@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930235369@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10736
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>;tag=as4b5bcf10
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0930235369@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1147847945 1147847945 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10736 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0930235369@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>;tag=as4b5bcf10
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>;tag=as4b5bcf10
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0930235369@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK1da41333;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as7502b5cf
To: <sip:0930235369@172.22.11.181>;tag=as4b5bcf10
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 07561b073b639599302ab7d716925671@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '07561b073b639599302ab7d716925671@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0938369923@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 15:58:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 250477340 250477340 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14962 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14962
Looking for 0938369923 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938369923@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938369923@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11336
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>;tag=as4bb571c1
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0938369923@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 679500721 679500721 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11336 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0938369923@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>;tag=as4bb571c1
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>;tag=as4bb571c1
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0938369923@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK72e89b9d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as323a143c
To: <sip:0938369923@172.22.11.181>;tag=as4bb571c1
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '696b55a40e2fd7aa52bcb7a36c80d4dc@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10128689;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:00:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 867990385 867990385 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14918 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14918
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10128689;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10128689;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10754
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10128689;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>;tag=as20cd9227
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1277127234 1277127234 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10754 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10754
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10128689;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>;tag=as20cd9227
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1277127234 1277127235 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10754 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK28bbcd10;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>;tag=as20cd9227
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3990afc7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>;tag=as20cd9227
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3990afc7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as73d42a64
To: <sip:0939011286@172.22.11.181>;tag=as20cd9227
Call-ID: 1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ce9d620;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:01:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 394664193 394664193 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10482 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10482
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ce9d620;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ce9d620;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16548
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ce9d620;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>;tag=as4d7b0786
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 591238352 591238352 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16548 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Really destroying SIP dialog '1ac31e742402751f1f8d9cf23e15831e@172.22.11.142:5060' Method: BYE
Audio is at 16548
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6ce9d620;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>;tag=as4d7b0786
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 591238352 591238353 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16548 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK256f2c0d;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>;tag=as4d7b0786
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK47b4145a;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>;tag=as4d7b0786
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK47b4145a;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as1b0e0077
To: <sip:0635714975@172.22.11.181>;tag=as4d7b0786
Call-ID: 59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '59f95a0a5d8f53b3735f2ec879e581cf@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0636393060@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6db39d88;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:06:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1388785430 1388785430 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19526 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19526
Looking for 0636393060 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6db39d88;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636393060@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6db39d88;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636393060@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15540
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6db39d88;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>;tag=as5a02f02f
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636393060@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1445080858 1445080858 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15540 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15540
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6db39d88;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>;tag=as5a02f02f
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0636393060@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1445080858 1445080859 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15540 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0636393060@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7d23adb3;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
To: <sip:0636393060@172.22.11.181>;tag=as5a02f02f
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935288746@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48c88242;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:06:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 672610137 672610137 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15190 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15190
Looking for 0935288746 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48c88242;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48c88242;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10952
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48c88242;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>;tag=as12ce4193
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 466558829 466558829 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10952 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Scheduling destruction of SIP dialog '1d5849cf628812a81b2677532a709183@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK001c272c;rport
Max-Forwards: 70
From: <sip:0636393060@172.22.11.181>;tag=as5a02f02f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK001c272c;received=172.22.11.181;rport=5060
From: <sip:0636393060@172.22.11.181>;tag=as5a02f02f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as54da5909
Call-ID: 1d5849cf628812a81b2677532a709183@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1d5849cf628812a81b2677532a709183@172.22.11.142:5060' Method: ACK
Audio is at 10952
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48c88242;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>;tag=as12ce4193
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935288746@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 466558829 466558830 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10952 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935288746@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK73dcc085;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
To: <sip:0935288746@172.22.11.181>;tag=as12ce4193
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52addba8;rport
Max-Forwards: 70
From: <sip:0935288746@172.22.11.181>;tag=as12ce4193
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK52addba8;received=172.22.11.181;rport=5060
From: <sip:0935288746@172.22.11.181>;tag=as12ce4193
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as4292b872
Call-ID: 3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3b860faf111b65296fc9e21a46b0b708@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0931043251@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK277ab678;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:34:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1615537225 1615537225 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19908 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19908
Looking for 0931043251 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK277ab678;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK277ab678;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 13254
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK277ab678;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>;tag=as6f73b5f3
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1092557462 1092557462 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13254 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 13254
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK277ab678;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>;tag=as6f73b5f3
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0931043251@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1092557462 1092557463 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 13254 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0931043251@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2945c6f6;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>;tag=as6f73b5f3
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7403d16b;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:35:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1934266434 1934266434 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19888 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19888
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7403d16b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7403d16b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18618
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7403d16b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>;tag=as5ea1d2c6
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1949668803 1949668803 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18618 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18618
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7403d16b;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>;tag=as5ea1d2c6
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1949668803 1949668804 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18618 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7917f539;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
To: <sip:0935183950@172.22.11.181>;tag=as5ea1d2c6
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d171540;rport
Max-Forwards: 70
From: <sip:0935183950@172.22.11.181>;tag=as5ea1d2c6
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK6d171540;received=172.22.11.181;rport=5060
From: <sip:0935183950@172.22.11.181>;tag=as5ea1d2c6
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3704f61f
Call-ID: 124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '124ac9e109fed5195a40c5531c70341d@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0931043251@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5ba362aa;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>;tag=as6f73b5f3
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5ba362aa;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as390a380d
To: <sip:0931043251@172.22.11.181>;tag=as6f73b5f3
Call-ID: 3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '3fe860cb7c3648f0677c12611e0871e3@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK775f78b7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:46:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 667060685 667060685 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 15310 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:15310
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK775f78b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK775f78b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10224
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK775f78b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>;tag=as5c5882c9
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1675085560 1675085560 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10224 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10224
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK775f78b7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>;tag=as5c5882c9
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1675085560 1675085561 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10224 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3b07aa55;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>;tag=as5c5882c9
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK106146b5;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>;tag=as5c5882c9
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK106146b5;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as0326c0b0
To: <sip:0935183950@172.22.11.181>;tag=as5c5882c9
Call-ID: 3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK09de5487;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:47:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1009645172 1009645172 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10350 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10350
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK09de5487;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK09de5487;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14262
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK09de5487;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>;tag=as6499a912
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 19532347 19532347 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14262 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14262
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK09de5487;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>;tag=as6499a912
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 19532347 19532348 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14262 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7f6b908f;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
To: <sip:0935183950@172.22.11.181>;tag=as6499a912
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3e4b01c16b17e42b3f26ab1159a1c50c@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK651e5ff4;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:51:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 127189349 127189349 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19110 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19110
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK651e5ff4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK651e5ff4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10478
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK651e5ff4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>;tag=as342b3fb2
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1417835045 1417835045 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10478 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10478
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK651e5ff4;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>;tag=as342b3fb2
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1417835045 1417835046 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10478 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK3ad27c20;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
To: <sip:0635714975@172.22.11.181>;tag=as342b3fb2
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47d4f2d5;rport
Max-Forwards: 70
From: <sip:0635714975@172.22.11.181>;tag=as342b3fb2
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47d4f2d5;received=172.22.11.181;rport=5060
From: <sip:0635714975@172.22.11.181>;tag=as342b3fb2
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as46efc707
Call-ID: 3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '3f5dccc70971f076173cfc672ab95978@172.22.11.142:5060' Method: ACK
Scheduling destruction of SIP dialog '11f518756b01be085139c5fb10e4b829@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47637974;rport
Max-Forwards: 70
From: <sip:0935183950@172.22.11.181>;tag=as6499a912
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK47637974;received=172.22.11.181;rport=5060
From: <sip:0935183950@172.22.11.181>;tag=as6499a912
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as682dba75
Call-ID: 11f518756b01be085139c5fb10e4b829@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '11f518756b01be085139c5fb10e4b829@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK14dbecd7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:53:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1262861073 1262861073 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 13316 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:13316
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK14dbecd7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK14dbecd7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15614
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK14dbecd7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>;tag=as4e913a2e
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1799657010 1799657010 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15614 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15614
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK14dbecd7;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>;tag=as4e913a2e
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1799657010 1799657011 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15614 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK5bfbe250;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>;tag=as4e913a2e
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK592e0805;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>;tag=as4e913a2e
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK592e0805;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3acd1b15
To: <sip:0935183950@172.22.11.181>;tag=as4e913a2e
Call-ID: 64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b8fafc0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 16:56:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 2111711635 2111711635 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 10322 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:10322
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b8fafc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b8fafc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 16476
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b8fafc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>;tag=as3f05d2ba
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1814831774 1814831774 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16476 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 16476
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0b8fafc0;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>;tag=as3f05d2ba
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1814831774 1814831775 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 16476 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0aa0974c;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>;tag=as3f05d2ba
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '64ee22a57d6cb58a0000505715495fa6@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6f23dc63;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>;tag=as3f05d2ba
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK6f23dc63;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as62d16478
To: <sip:0935183950@172.22.11.181>;tag=as3f05d2ba
Call-ID: 6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '6f981ecc6a8501ac1287755974daa364@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b9e8402;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 17:13:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 517960179 517960179 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18834 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18834
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b9e8402;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b9e8402;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14096
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b9e8402;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>;tag=as560095e4
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1509279901 1509279901 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14096 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 14096
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK2b9e8402;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>;tag=as560095e4
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1509279901 1509279902 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14096 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK031aced0;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>;tag=as560095e4
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0639133481@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK481af4aa;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 17:13:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1262598495 1262598495 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 14440 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:14440
Looking for 0639133481 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK481af4aa;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0639133481@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK481af4aa;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0639133481@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 18714
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK481af4aa;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>;tag=as299011a4
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0639133481@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 865172110 865172110 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18714 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 18714
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK481af4aa;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>;tag=as299011a4
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0639133481@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 865172110 865172111 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 18714 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0639133481@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK394fd729;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>;tag=as299011a4
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0639133481@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fef579e;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>;tag=as299011a4
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '6c16ffa471423250610ba82c58b797de@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7fef579e;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as74a9f41b
To: <sip:0639133481@172.22.11.181>;tag=as299011a4
Call-ID: 6c16ffa471423250610ba82c58b797de@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '6c16ffa471423250610ba82c58b797de@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
BYE sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK54a91672;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>;tag=as560095e4
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)
Scheduling destruction of SIP dialog '340a2b6300c18db934920d940d3497e2@172.22.11.142:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK54a91672;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as3cdbb754
To: <sip:0935183950@172.22.11.181>;tag=as560095e4
Call-ID: 340a2b6300c18db934920d940d3497e2@172.22.11.142:5060
CSeq: 103 BYE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '340a2b6300c18db934920d940d3497e2@172.22.11.142:5060' Method: BYE

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK67128bc8;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 17:24:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 861062203 861062203 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19544 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19544
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK67128bc8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK67128bc8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 12894
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK67128bc8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>;tag=as76579a9f
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 565739889 565739889 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12894 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 12894
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK67128bc8;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>;tag=as76579a9f
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 565739889 565739890 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 12894 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK48ac2ba7;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
To: <sip:0935183950@172.22.11.181>;tag=as76579a9f
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '330e2469536ce52d060f25a06a18b571@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK695dd00a;rport
Max-Forwards: 70
From: <sip:0935183950@172.22.11.181>;tag=as76579a9f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK695dd00a;received=172.22.11.181;rport=5060
From: <sip:0935183950@172.22.11.181>;tag=as76579a9f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as05ad1489
Call-ID: 330e2469536ce52d060f25a06a18b571@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '330e2469536ce52d060f25a06a18b571@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0935183950@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c68c866;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 17:29:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 82439938 82439938 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 17716 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:17716
Looking for 0935183950 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c68c866;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c68c866;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 10854
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c68c866;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>;tag=as721c1062
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 464155525 464155525 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10854 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 10854
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK7c68c866;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>;tag=as721c1062
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0935183950@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 464155525 464155526 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 10854 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0935183950@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK10d11d49;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
To: <sip:0935183950@172.22.11.181>;tag=as721c1062
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK0c7dc404;rport
Max-Forwards: 70
From: <sip:0935183950@172.22.11.181>;tag=as721c1062
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK0c7dc404;received=172.22.11.181;rport=5060
From: <sip:0935183950@172.22.11.181>;tag=as721c1062
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as5f114809
Call-ID: 0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '0136b4a21af19f176cb4850c76f74013@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 18:57:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 753852406 753852406 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 19952 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:19952
Looking for 0635714975 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 14976
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>;tag=as59aad478
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0635714975@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1400214793 1400214793 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 14976 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
CANCEL sip:0635714975@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.22.11.142:5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>;tag=as59aad478
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>;tag=as59aad478
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 CANCEL
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0635714975@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK593363a1;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as43fccfad
To: <sip:0635714975@172.22.11.181>;tag=as59aad478
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7ccf3c976ccec9685dc8bbef753dff77@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK402d9474;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 20:35:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1044655745 1044655745 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 16302 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:16302
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK402d9474;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK402d9474;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 15498
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK402d9474;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>;tag=as54981b4f
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2038824437 2038824437 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15498 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 15498
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK402d9474;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>;tag=as54981b4f
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2038824437 2038824438 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 15498 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK0473f308;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
To: <sip:0939011286@172.22.11.181>;tag=as54981b4f
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK4f87554b;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as54981b4f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK4f87554b;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as54981b4f
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as56d1e59a
Call-ID: 0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '0c9150f90fc19d4952f83879720b5289@172.22.11.142:5060' Method: ACK

<--- SIP read from UDP:172.22.11.142:5060 --->
INVITE sip:0939011286@172.22.11.181 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08c28486;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 25 Jul 2015 20:39:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1377088253 1377088253 IN IP4 172.22.11.142
s=Asterisk PBX 1.8.26.1
c=IN IP4 172.22.11.142
t=0 0
m=audio 18668 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Sending to 172.22.11.142:5060 (NAT)
Using INVITE request as basis request - 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
peer=(null).
No matching peer for '272-172.22.11.142' from '172.22.11.142:5060'
user_and_ip=272-172.22.11.142, res=-100, sipmethod=5.
peer=(null).
peer=172.22.11.142.
Found peer '172.22.11.142' for '272' from 172.22.11.142:5060
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.22.11.142:18668
Looking for 0939011286 in sipinbound (domain 172.22.11.181)
list_route: hop: <sip:272@172.22.11.142:5060>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08c28486;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08c28486;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Length: 0


<------------>
Audio is at 11446
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08c28486;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>;tag=as2e36b4c2
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2146911503 2146911503 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Audio is at 11446
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK08c28486;received=172.22.11.142;rport=5060
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>;tag=as2e36b4c2
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 INVITE
Server: VoxStack Wireless Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0939011286@172.22.11.181:5060>
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 2146911503 2146911504 IN IP4 172.22.11.181
s=VoxStack Wireless Gateway
c=IN IP4 172.22.11.181
t=0 0
m=audio 11446 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:172.22.11.142:5060 --->
ACK sip:0939011286@172.22.11.181:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.142:5060;branch=z9hG4bK37968aa2;rport
Max-Forwards: 70
From: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
To: <sip:0939011286@172.22.11.181>;tag=as2e36b4c2
Contact: <sip:272@172.22.11.142:5060>
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:272@172.22.11.142:5060> for address/port to send to
set_destination: set destination to 172.22.11.142:5060
Reliably Transmitting (no NAT) to 172.22.11.142:5060:
BYE sip:272@172.22.11.142:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK231661e4;rport
Max-Forwards: 70
From: <sip:0939011286@172.22.11.181>;tag=as2e36b4c2
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 BYE
User-Agent: VoxStack Wireless Gateway
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


---

<--- SIP read from UDP:172.22.11.142:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.11.181:5060;branch=z9hG4bK231661e4;received=172.22.11.181;rport=5060
From: <sip:0939011286@172.22.11.181>;tag=as2e36b4c2
To: "Dispetcher 2" <sip:272@172.22.11.142>;tag=as45619c13
Call-ID: 1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060
CSeq: 102 BYE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1c4223a1510b6f5863b72d596062ee49@172.22.11.142:5060' Method: ACK
