AstT*CLI> sip set debug on
SIP Debugging enabled
[2015-09-09 09:23:23] 
<--- SIP read from UDP:172.16.22.3:5060 --->
INVITE sip:93297121@NVMSSIP;user=phone SIP/2.0
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>
Max-Forwards: 70
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000005930994266154
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63457 INVITE
P-Asserted-Identity: <sip:+37493666666@172.16.22.3;user=phone>
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
P-Charging-Vector: icid-value=9DB7FF2000-0909-09232204;icid-generated-at=172.16.22.3;orig-ioi=MSC2VIVA
Supported: 100rel
Content-Type: application/sdp
Contact: <sip:172.16.22.3:5060;transport=UDP>
Content-Length: 365

v=0
o=- 10812701 10812701 IN IP4 172.16.22.3
s=-
c=IN IP4 172.16.22.7
t=0 0
a=sendrecv
m=audio 16958 RTP/AVP 8 0 18 96 97
c=IN IP4 172.16.22.7
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 G729/8000
a=fmtp:96 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40
<------------->
[2015-09-09 09:23:23] --- (15 headers 19 lines) ---
[2015-09-09 09:23:23] Sending to 172.16.22.3:5060 (NAT)
[2015-09-09 09:23:23] Using INVITE request as basis request - qJcQ1332929090501-AAAACHFH-@172.16.22.3
[2015-09-09 09:23:23] Found peer 'msco' for '+37493666666' from 172.16.22.3:5060
[2015-09-09 09:23:23]   == Using SIP RTP TOS bits 184
[2015-09-09 09:23:23]   == Using SIP RTP CoS mark 5
[2015-09-09 09:23:23] Found RTP audio format 8
[2015-09-09 09:23:23] Found RTP audio format 0
[2015-09-09 09:23:23] Found RTP audio format 18
[2015-09-09 09:23:23] Found RTP audio format 96
[2015-09-09 09:23:23] Found RTP audio format 97
[2015-09-09 09:23:23] Found audio description format PCMA for ID 8
[2015-09-09 09:23:23] Found audio description format PCMU for ID 0
[2015-09-09 09:23:23] Found audio description format G729 for ID 18
[2015-09-09 09:23:23] Found audio description format G729 for ID 96
[2015-09-09 09:23:23] Found audio description format telephone-event for ID 97
[2015-09-09 09:23:23] Capabilities: us - (alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
[2015-09-09 09:23:23] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-09-09 09:23:23] Peer audio RTP is at port 172.16.22.7:16958
[2015-09-09 09:23:23] Looking for 93297121 in from_msc (domain NVMSSIP)
[2015-09-09 09:23:23] list_route: hop: <sip:172.16.22.3:5060;transport=UDP>
[2015-09-09 09:23:23] 
<--- Transmitting (NAT) to 172.16.22.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000005930994266154;received=172.16.22.3;rport=5060
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63457 INVITE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:93297121@172.16.22.19:5060>
Content-Length: 0


<------------>
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:1] NoOp("SIP/msco-000000b7", "ER") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:2] Set("SIP/msco-000000b7", "Diversion=374") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:3] Set("SIP/msco-000000b7", "Diversion=37493297121") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:4] NoOp("SIP/msco-000000b7", "div=37493297121 callerid=+37493666666") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:5] GotoIf("SIP/msco-000000b7", "0?testb") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:6] GotoIf("SIP/msco-000000b7", "1?viva_numbers") in new stack
[2015-09-09 09:23:23]     -- Goto (from_msc,93297121,22)
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:22] NoOp("SIP/msco-000000b7", "37493297121") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:23] NoOp("SIP/msco-000000b7", "CALLERID(num)=+37493666666") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:24] Set("SIP/msco-000000b7", "CALLERID(num)=37493666666") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:25] Set("SIP/msco-000000b7", "privacy=") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:26] NoOp("SIP/msco-000000b7", "37493666666") in new stack
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:27] GotoIf("SIP/msco-000000b7", "1?testa") in new stack
[2015-09-09 09:23:23]     -- Goto (from_msc,93297121,38)
[2015-09-09 09:23:23]     -- Executing [93297121@from_msc:38] Goto("SIP/msco-000000b7", "localoca,37493297121,1") in new stack
[2015-09-09 09:23:23]     -- Goto (localoca,37493297121,1)
[2015-09-09 09:23:23]     -- Executing [37493297121@localoca:1] NoOp("SIP/msco-000000b7", "37493297121") in new stack
[2015-09-09 09:23:23]     -- Executing [37493297121@localoca:2] Read("SIP/msco-000000b7", "SWITCH,beep,1") in new stack
[2015-09-09 09:23:23]     -- Accepting a maximum of 1 digits.
[2015-09-09 09:23:23] Audio is at 11168
[2015-09-09 09:23:23] Adding codec 100004 (alaw) to SDP
[2015-09-09 09:23:23] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-09 09:23:23] 
<--- Reliably Transmitting (NAT) to 172.16.22.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000005930994266154;received=172.16.22.3;rport=5060
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>;tag=as3ad5b49f
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63457 INVITE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:93297121@172.16.22.19:5060>
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 792083395 792083395 IN IP4 172.16.22.19
s=Asterisk PBX 10.1.2
c=IN IP4 172.16.22.19
t=0 0
m=audio 11168 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[2015-09-09 09:23:23] 
<--- SIP read from UDP:172.16.22.3:5060 --->
ACK sip:93297121@172.16.22.19:5060 SIP/2.0
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>;tag=as3ad5b49f
Max-Forwards: 70
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000070404693657845
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63457 ACK
Content-Length: 0

<------------->
[2015-09-09 09:23:23] --- (8 headers 0 lines) ---
[2015-09-09 09:23:23]     -- <SIP/msco-000000b7> Playing 'beep.gsm' (language 'en')
[2015-09-09 09:23:24]     -- User entered '2'
[2015-09-09 09:23:24]     -- Executing [37493297121@localoca:3] GotoIf("SIP/msco-000000b7", "0?m") in new stack
[2015-09-09 09:23:24]     -- Executing [37493297121@localoca:4] GotoIf("SIP/msco-000000b7", "1?a") in new stack
[2015-09-09 09:23:24]     -- Goto (localoca,37493297121,12)
[2015-09-09 09:23:24]     -- Executing [37493297121@localoca:12] NoOp("SIP/msco-000000b7", "37493297121") in new stack
[2015-09-09 09:23:24]     -- Executing [37493297121@localoca:13] Set("SIP/msco-000000b7", "CALLERID(num)=79255812464") in new stack
[2015-09-09 09:23:24]     -- Executing [37493297121@localoca:14] Dial("SIP/msco-000000b7", "SIP/multifon-out/+37493298888,60") in new stack
[2015-09-09 09:23:24]   == Using SIP RTP TOS bits 184
[2015-09-09 09:23:24]   == Using SIP RTP CoS mark 5
[2015-09-09 09:23:24] Audio is at 18448
[2015-09-09 09:23:24] Adding codec 100004 (alaw) to SDP
[2015-09-09 09:23:24] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-09 09:23:24] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:+37493298888@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP 217.74.3.171:5060;branch=z9hG4bK14326523;rport
Max-Forwards: 70
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@193.201.229.35>
Contact: <sip:79255812464@217.74.3.171:5060>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 09 Sep 2015 05:23:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1438789579 1438789579 IN IP4 217.74.3.171
s=Asterisk PBX 10.1.2
c=IN IP4 217.74.3.171
t=0 0
m=audio 18448 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2015-09-09 09:23:24]     -- Called SIP/multifon-out/+37493298888
[2015-09-09 09:23:24] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK14326523;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 102 INVITE

<------------->
[2015-09-09 09:23:24] --- (6 headers 0 lines) ---
[2015-09-09 09:23:24] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK14326523;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>;tag=SD95nm899-55EC32463135364161C16502
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 102 INVITE
Proxy-Authenticate: Digest nonce="MTQ0MTc3NjIwNDoNokoWfCq9PXLNSKM9B80I",opaque="MTQ0MTc3NjIwNDoNokoWfCq9PXLNSKM9B80I",algorithm=md5,realm="BREDBAND",qop="auth"
Reason: SEM;cause=5;text="Need auth"
Content-Length: 0

<------------->
[2015-09-09 09:23:24] --- (9 headers 0 lines) ---
[2015-09-09 09:23:24] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:+37493298888@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP 217.74.3.171:5060;branch=z9hG4bK14326523;rport
Max-Forwards: 70
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@193.201.229.35>;tag=SD95nm899-55EC32463135364161C16502
Contact: <sip:79255812464@217.74.3.171:5060>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-09 09:23:24] Audio is at 18448
[2015-09-09 09:23:24] Adding codec 100004 (alaw) to SDP
[2015-09-09 09:23:24] Adding non-codec 0x1 (telephone-event) to SDP
[2015-09-09 09:23:24] Reliably Transmitting (NAT) to 193.201.229.35:5060:
INVITE sip:+37493298888@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP 217.74.3.171:5060;branch=z9hG4bK3ecf7f41;rport
Max-Forwards: 70
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@193.201.229.35>
Contact: <sip:79255812464@217.74.3.171:5060>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.1.2
Proxy-Authorization: Digest username="79255812464", realm="BREDBAND", algorithm=MD5, uri="sip:+37493298888@193.201.229.35", nonce="MTQ0MTc3NjIwNDoNokoWfCq9PXLNSKM9B80I", response="f3e12be545b9d1b4848542bae50f033d", opaque="MTQ0MTc3NjIwNDoNokoWfCq9PXLNSKM9B80I", qop=auth, cnonce="2bcd6a19", nc=00000001
Date: Wed, 09 Sep 2015 05:23:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1438789579 1438789580 IN IP4 217.74.3.171
s=Asterisk PBX 10.1.2
c=IN IP4 217.74.3.171
t=0 0
m=audio 18448 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[2015-09-09 09:23:24] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK3ecf7f41;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 INVITE

<------------->
[2015-09-09 09:23:24] --- (6 headers 0 lines) ---

<------------->
[2015-09-09 09:23:27] --- (14 headers 0 lines) ---
[2015-09-09 09:23:27] Really destroying SIP dialog '01c5a9172497bc413731502c0d484164@217.74.3.171:5060' Method: OPTIONS
[2015-09-09 09:23:31] Really destroying SIP dialog 'iGfS3125929090501-AAAAAGKN-@172.16.22.3' Method: BYE
[2015-09-09 09:23:32] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK3ecf7f41;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>;tag=SD95nm899-C2B03246313536416BC16502
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 INVITE
Allow: OPTIONS,CANCEL,UPDATE
Content-Length: 0
Contact: <sip:+37493298888@193.201.229.35:5060;transport=udp>

<------------->
[2015-09-09 09:23:32] --- (9 headers 0 lines) ---
[2015-09-09 09:23:32] list_route: hop: <sip:+37493298888@193.201.229.35:5060;transport=udp>
[2015-09-09 09:23:32]     -- SIP/multifon-out-000000b8 is ringing
[2015-09-09 09:23:43] Reliably Transmitting (NAT) to 172.16.21.92:5060:
OPTIONS sip:172.16.21.92 SIP/2.0
Via: SIP/2.0/UDP 172.16.21.103:5060;branch=z9hG4bK49f025f7;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.16.21.103>;tag=as1a579705
To: <sip:172.16.21.92>
Contact: <sip:Unknown@172.16.21.103:5060>
Call-ID: 5c8a913e459273d6741a94ed091e9f05@172.16.21.103:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.1.2
Date: Wed, 09 Sep 2015 05:23:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[2015-09-09 09:23:43] 
<--- SIP read from UDP:172.16.21.92:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.21.103:5060;branch=z9hG4bK49f025f7;received=172.16.21.103;rport=5060
From: "Unknown" <sip:Unknown@172.16.21.103>;tag=as1a579705
To: <sip:172.16.21.92>;tag=as4cbedeaf
Call-ID: 5c8a913e459273d6741a94ed091e9f05@172.16.21.103:5060
CSeq: 102 OPTIONS
Server: rsa-vis
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:172.16.21.92:5060>
Accept: application/sdp
Content-Length: 0

<------------->
[2015-09-09 09:23:43] --- (12 headers 0 lines) ---
[2015-09-09 09:23:43] Really destroying SIP dialog '5c8a913e459273d6741a94ed091e9f05@172.16.21.103:5060' Method: OPTIONS
[2015-09-09 09:23:45] 
<--- SIP read from UDP:172.16.22.3:5060 --->
BYE sip:93297121@172.16.22.19:5060 SIP/2.0
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>;tag=as3ad5b49f
Max-Forwards: 70
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000001095578105670
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63458 BYE
Content-Length: 0

<------------->
[2015-09-09 09:23:45] --- (8 headers 0 lines) ---
[2015-09-09 09:23:45] Sending to 172.16.22.3:5060 (NAT)
[2015-09-09 09:23:45] Scheduling destruction of SIP dialog 'qJcQ1332929090501-AAAACHFH-@172.16.22.3' in 32000 ms (Method: BYE)
[2015-09-09 09:23:45] 
<--- Transmitting (NAT) to 172.16.22.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.22.3:5060;branch=z9hG4bK00000001095578105670;received=172.16.22.3;rport=5060
From: <sip:+37493666666@172.16.22.3;user=phone>;tag=1189127718
To: <sip:93297121@NVMSSIP;user=phone>;tag=as3ad5b49f
Call-ID: qJcQ1332929090501-AAAACHFH-@172.16.22.3
CSeq: 63458 BYE
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[2015-09-09 09:23:45] Scheduling destruction of SIP dialog '251437251a1601c54a9059541e0fd9b6@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-09 09:23:45] set_destination: Parsing <sip:+37493298888@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-09 09:23:45] set_destination: set destination to 193.201.229.35:5060
[2015-09-09 09:23:45] Reliably Transmitting (NAT) to 193.201.229.35:5060:
CANCEL sip:+37493298888@193.201.229.35 SIP/2.0
Via: SIP/2.0/UDP 217.74.3.171:5060;branch=z9hG4bK3ecf7f41;rport
Max-Forwards: 70
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@193.201.229.35>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-09 09:23:45] Scheduling destruction of SIP dialog '251437251a1601c54a9059541e0fd9b6@multifon.ru' in 6400 ms (Method: INVITE)
[2015-09-09 09:23:45]   == Spawn extension (localoca, 37493297121, 14) exited non-zero on 'SIP/msco-000000b7'
[2015-09-09 09:23:45] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK3ecf7f41;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>;tag=SD95nm899-C2B03246313536416BC16502
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 CANCEL

<------------->
[2015-09-09 09:23:45] --- (6 headers 0 lines) ---
[2015-09-09 09:23:45] 
<--- SIP read from UDP:193.201.229.35:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 217.74.3.171:5060;received=217.74.3.171;branch=z9hG4bK3ecf7f41;rport=5060
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@multifon.ru>;tag=SD95nm899-C2B03246313536416BC16502
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 INVITE
Content-Length: 0

<------------->
[2015-09-09 09:23:45] --- (7 headers 0 lines) ---
[2015-09-09 09:23:45] set_destination: Parsing <sip:+37493298888@193.201.229.35:5060;transport=udp> for address/port to send to
[2015-09-09 09:23:45] set_destination: set destination to 193.201.229.35:5060
[2015-09-09 09:23:45] Transmitting (NAT) to 193.201.229.35:5060:
ACK sip:+37493298888@193.201.229.35:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 217.74.3.171:5060;branch=z9hG4bK3ecf7f41;rport
Max-Forwards: 70
From: "79255812464" <sip:79255812464@multifon.ru>;tag=as0fbdfca0
To: <sip:+37493298888@193.201.229.35>;tag=SD95nm899-C2B03246313536416BC16502
Contact: <sip:79255812464@217.74.3.171:5060>
Call-ID: 251437251a1601c54a9059541e0fd9b6@multifon.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.1.2
Content-Length: 0


---
[2015-09-09 09:23:45] Really destroying SIP dialog '251437251a1601c54a9059541e0fd9b6@multifon.ru' Method: INVITE
[2015-09-09 09:23:46] 
<--- SIP read from UDP:172.16.21.92:5060 --->
OPTIONS sip:s@172.16.21.103:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.21.92:5060;branch=z9hG4bK57fab6a2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.21.92>;tag=as31a1ce56
To: <sip:s@172.16.21.103:5060>
Contact: <sip:asterisk@172.16.21.92:5060>
Call-ID: 56f3a9df38dbd96b377f845d4355753d@172.16.21.92:5060
CSeq: 102 OPTIONS
User-Agent: rsa-vis
Date: Wed, 09 Sep 2015 05:23:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
[2015-09-09 09:23:46] --- (13 headers 0 lines) ---
[2015-09-09 09:23:46] Looking for s in from-sip-external (domain 172.16.21.103)
[2015-09-09 09:23:46] 
<--- Transmitting (NAT) to 172.16.21.92:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.21.92:5060;branch=z9hG4bK57fab6a2;received=172.16.21.92;rport=5060
From: "asterisk" <sip:asterisk@172.16.21.92>;tag=as31a1ce56
To: <sip:s@172.16.21.103:5060>;tag=as3309a376
Call-ID: 56f3a9df38dbd96b377f845d4355753d@172.16.21.92:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 10.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:172.16.21.103:5060>
Accept: application/sdp
Content-Length: 0


<------------>
[2015-09-09 09:23:46] Scheduling destruction of SIP dialog '56f3a9df38dbd96b377f845d4355753d@172.16.21.92:5060' in 32000 ms (Method: OPTIONS)
AstT*CLI>