<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKb40a28ef94cb3e47
From: "611" <sip:611@10.1.1.220>;tag=cc956ba2-700753
To: "611" <sip:611@10.1.1.220>
Call-ID: 1B12-1220-466848125E5C7B2412BD-001@SipHost
CSeq:58 REGISTER
Contact: <sip:611@10.1.1.15:5060>
Expires:600
Max-Forwards:70
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKb40a28ef94cb3e47;received=10.1.1.15
From: "611" <sip:611@10.1.1.220>;tag=cc956ba2-700753
To: "611" <sip:611@10.1.1.220>;tag=as37051ca3
Call-ID: 1B12-1220-466848125E5C7B2412BD-001@SipHost
CSeq: 58 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d82480b"
Content-Length: 0


<------------>
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-466848125E5C7B2412BD-001@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK7893225fa1dd7205
From: "612" <sip:612@10.1.1.220>;tag=29bb8c4d-700753
To: "612" <sip:612@10.1.1.220>
Call-ID: 1B12-1220-466848122EBAD76C7494-002@SipHost
CSeq:70 REGISTER
Contact: <sip:612@10.1.1.15:5060>
Expires:600
Max-Forwards:70
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK7893225fa1dd7205;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=29bb8c4d-700753
To: "612" <sip:612@10.1.1.220>;tag=as3f477009
Call-ID: 1B12-1220-466848122EBAD76C7494-002@SipHost
CSeq: 70 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="350e9230"
Content-Length: 0


<------------>
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-466848122EBAD76C7494-002@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKa755e3ca4d3c8f4a
From: "611" <sip:611@10.1.1.220>;tag=cc956ba2-700753
To: "611" <sip:611@10.1.1.220>
Call-ID: 1B12-1220-466848125E5C7B2412BD-001@SipHost
CSeq:59 REGISTER
Contact: <sip:611@10.1.1.15:5060>
Expires:600
Max-Forwards:70
Authorization:Digest username="611",realm="asterisk",nonce="4d82480b",uri="sip:10.1.1.220:5060",response="785c1da040d06bd62fa6af8f1e89218d",algorithm=MD5
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKa755e3ca4d3c8f4a;received=10.1.1.15
From: "611" <sip:611@10.1.1.220>;tag=cc956ba2-700753
To: "611" <sip:611@10.1.1.220>;tag=as37051ca3
Call-ID: 1B12-1220-466848125E5C7B2412BD-001@SipHost
CSeq: 59 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:611@10.1.1.15:5060>;expires=600
Date: Mon, 20 Nov 2017 08:30:43 GMT
Content-Length: 0


<------------>
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-466848125E5C7B2412BD-001@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK36df737aa96059b2
From: "612" <sip:612@10.1.1.220>;tag=29bb8c4d-700753
To: "612" <sip:612@10.1.1.220>
Call-ID: 1B12-1220-466848122EBAD76C7494-002@SipHost
CSeq:71 REGISTER
Contact: <sip:612@10.1.1.15:5060>
Expires:600
Max-Forwards:70
Authorization:Digest username="612",realm="asterisk",nonce="350e9230",uri="sip:10.1.1.220:5060",response="469d005b59fbf5bae8ae63909739babd",algorithm=MD5
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK36df737aa96059b2;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=29bb8c4d-700753
To: "612" <sip:612@10.1.1.220>;tag=as3f477009
Call-ID: 1B12-1220-466848122EBAD76C7494-002@SipHost
CSeq: 71 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:612@10.1.1.15:5060>;expires=600
Date: Mon, 20 Nov 2017 08:30:43 GMT
Content-Length: 0


<------------>
[Nov 20 10:30:43] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-466848122EBAD76C7494-002@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:30:50] NOTICE[100192] chan_sip.c:    -- Re-registration for  3333333@12.34.56.78
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21da157d
Max-Forwards: 70
From: <sip:3333333@12.34.56.78>;tag=as1e786d1e
To: <sip:3333333@12.34.56.78>
Call-ID: 34f477d60347f76444326a3628cf26be@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="3333333", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="03814c6f", response="8537d5616b04fcae19e5f7e5874a404d"
Expires: 120
Contact: <sip:3333333@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:50] NOTICE[100192] chan_sip.c:    -- Re-registration for  8888888@12.34.56.78
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK70cffa01
Max-Forwards: 70
From: <sip:8888888@12.34.56.78>;tag=as79b4259e
To: <sip:8888888@12.34.56.78>
Call-ID: 7a32e8bb4f29862c7e1648184cd997c2@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="8888888", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="2c3081ec", response="398fe2c6e11d30dfa86c51d47128328d"
Expires: 120
Contact: <sip:8888888@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21da157d;received=87.65.43.21
From: <sip:3333333@12.34.56.78>;tag=as1e786d1e
To: <sip:3333333@12.34.56.78>;tag=as56795f31
Call-ID: 34f477d60347f76444326a3628cf26be@10.1.1.220
CSeq: 410 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d6e5e44"
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 12.34.56.78
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK3437cf70
Max-Forwards: 70
From: <sip:3333333@12.34.56.78>;tag=as1e786d1e
To: <sip:3333333@12.34.56.78>
Call-ID: 34f477d60347f76444326a3628cf26be@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="3333333", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="6d6e5e44", response="766fe08889a1a3032207fc174b60f878"
Expires: 120
Contact: <sip:3333333@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK70cffa01;received=87.65.43.21
From: <sip:8888888@12.34.56.78>;tag=as79b4259e
To: <sip:8888888@12.34.56.78>;tag=as11ef8a9a
Call-ID: 7a32e8bb4f29862c7e1648184cd997c2@10.1.1.220
CSeq: 410 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d11b0f8"
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 12.34.56.78
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK467703f7
Max-Forwards: 70
From: <sip:8888888@12.34.56.78>;tag=as79b4259e
To: <sip:8888888@12.34.56.78>
Call-ID: 7a32e8bb4f29862c7e1648184cd997c2@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="8888888", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="2d11b0f8", response="6f6b0f980f4916355c0f02d23b43afc0"
Expires: 120
Contact: <sip:8888888@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:3333333@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK501fe6be;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as5dc3327d
To: <sip:3333333@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 5fce3cd43bf064b2333fd4f307b52b3f@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:30:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Looking for 3333333 in public (domain 87.65.43.21)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK501fe6be;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as5dc3327d
To: <sip:3333333@87.65.43.21:5060>;tag=as511cb715
Call-ID: 5fce3cd43bf064b2333fd4f307b52b3f@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '5fce3cd43bf064b2333fd4f307b52b3f@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK3437cf70;received=87.65.43.21
From: <sip:3333333@12.34.56.78>;tag=as1e786d1e
To: <sip:3333333@12.34.56.78>;tag=as56795f31
Call-ID: 34f477d60347f76444326a3628cf26be@10.1.1.220
CSeq: 411 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:3333333@87.65.43.21:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:52 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:50] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 12.34.56.78 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '34f477d60347f76444326a3628cf26be@10.1.1.220' Method: REGISTER
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:8888888@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK53b3c7c9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as3c61aca0
To: <sip:8888888@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 6e3565174e1679496ea3ccdb0432e37f@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:30:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Looking for 8888888 in public (domain 87.65.43.21)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK53b3c7c9;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as3c61aca0
To: <sip:8888888@87.65.43.21:5060>;tag=as3c319988
Call-ID: 6e3565174e1679496ea3ccdb0432e37f@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '6e3565174e1679496ea3ccdb0432e37f@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK467703f7;received=87.65.43.21
From: <sip:8888888@12.34.56.78>;tag=as79b4259e
To: <sip:8888888@12.34.56.78>;tag=as11ef8a9a
Call-ID: 7a32e8bb4f29862c7e1648184cd997c2@10.1.1.220
CSeq: 411 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:8888888@87.65.43.21:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:52 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:50] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 12.34.56.78 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '7a32e8bb4f29862c7e1648184cd997c2@10.1.1.220' Method: REGISTER
[Nov 20 10:30:50] NOTICE[100192] chan_sip.c:    -- Re-registration for  6666666@12.34.56.78
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:50] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK5c7c9cb9
Max-Forwards: 70
From: <sip:6666666@12.34.56.78>;tag=as4fa91fc0
To: <sip:6666666@12.34.56.78>
Call-ID: 5a2b96491ca07109684d975653f12a89@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="6666666", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="53078cd6", response="0ec4f7d54b474ef5918c3ffa6794c78b"
Expires: 120
Contact: <sip:6666666@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK5c7c9cb9;received=87.65.43.21
From: <sip:6666666@12.34.56.78>;tag=as4fa91fc0
To: <sip:6666666@12.34.56.78>;tag=as0870b026
Call-ID: 5a2b96491ca07109684d975653f12a89@10.1.1.220
CSeq: 410 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="412da26d"
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 12.34.56.78
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK7a675be7
Max-Forwards: 70
From: <sip:6666666@12.34.56.78>;tag=as4fa91fc0
To: <sip:6666666@12.34.56.78>
Call-ID: 5a2b96491ca07109684d975653f12a89@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="6666666", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="412da26d", response="adf45e8b993c969a918be83c3adb0a30"
Expires: 120
Contact: <sip:6666666@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:6666666@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK51e236f3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as19ded3b1
To: <sip:6666666@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 14a823a1033d53420110c2a460e36da1@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:30:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Looking for 6666666 in public (domain 87.65.43.21)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK51e236f3;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as19ded3b1
To: <sip:6666666@87.65.43.21:5060>;tag=as44eee290
Call-ID: 14a823a1033d53420110c2a460e36da1@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '14a823a1033d53420110c2a460e36da1@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK7a675be7;received=87.65.43.21
From: <sip:6666666@12.34.56.78>;tag=as4fa91fc0
To: <sip:6666666@12.34.56.78>;tag=as0870b026
Call-ID: 5a2b96491ca07109684d975653f12a89@10.1.1.220
CSeq: 411 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:6666666@87.65.43.21:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:52 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 12.34.56.78 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '5a2b96491ca07109684d975653f12a89@10.1.1.220' Method: REGISTER
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c:    -- Re-registration for  7777777@12.34.56.78
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK1ddb05a8
Max-Forwards: 70
From: <sip:7777777@12.34.56.78>;tag=as530a70ee
To: <sip:7777777@12.34.56.78>
Call-ID: 71fd5ceb445111c23cd509fa5ed3266d@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="7777777", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="12e67dd9", response="ee1ff2906fa564e64a32c785a73a9618"
Expires: 120
Contact: <sip:7777777@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK1ddb05a8;received=87.65.43.21
From: <sip:7777777@12.34.56.78>;tag=as530a70ee
To: <sip:7777777@12.34.56.78>;tag=as2a367cf6
Call-ID: 71fd5ceb445111c23cd509fa5ed3266d@10.1.1.220
CSeq: 410 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c74833b"
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 12.34.56.78
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
REGISTER sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK46252906
Max-Forwards: 70
From: <sip:7777777@12.34.56.78>;tag=as530a70ee
To: <sip:7777777@12.34.56.78>
Call-ID: 71fd5ceb445111c23cd509fa5ed3266d@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="7777777", realm="asterisk", algorithm=MD5, uri="sip:12.34.56.78", nonce="6c74833b", response="e77fea68f741a6fc366087b75bdf0c17"
Expires: 120
Contact: <sip:7777777@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:7777777@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK5c63a96c;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as7c199608
To: <sip:7777777@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 11c7aa675bcea45e71260fd82ce40c60@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:30:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Looking for 7777777 in public (domain 87.65.43.21)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK5c63a96c;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as7c199608
To: <sip:7777777@87.65.43.21:5060>;tag=as132b6cdb
Call-ID: 11c7aa675bcea45e71260fd82ce40c60@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '11c7aa675bcea45e71260fd82ce40c60@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK46252906;received=87.65.43.21
From: <sip:7777777@12.34.56.78>;tag=as530a70ee
To: <sip:7777777@12.34.56.78>;tag=as2a367cf6
Call-ID: 71fd5ceb445111c23cd509fa5ed3266d@10.1.1.220
CSeq: 411 REGISTER
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:7777777@87.65.43.21:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:52 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 12.34.56.78 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '71fd5ceb445111c23cd509fa5ed3266d@10.1.1.220' Method: REGISTER
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c:    -- Re-registration for  9999999999@10.1.1.6
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
REGISTER sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK1aff5e6d
Max-Forwards: 70
From: <sip:9999999999@10.1.1.6>;tag=as399f2e25
To: <sip:9999999999@10.1.1.6>
Call-ID: 0ab72dd24cc63eda7962f1922717ab1d@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="9999999999", realm="asterisk", algorithm=MD5, uri="sip:10.1.1.6", nonce="769526ad", response="e66f2a2bc789fa6b75e9704663ef5241"
Expires: 120
Contact: <sip:9999999999@10.1.1.220:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK1aff5e6d;received=10.1.1.220
From: <sip:9999999999@10.1.1.6>;tag=as399f2e25
To: <sip:9999999999@10.1.1.6>;tag=as30f7a95a
Call-ID: 0ab72dd24cc63eda7962f1922717ab1d@10.1.1.220
CSeq: 410 REGISTER
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3315f650"
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 10.1.1.6
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
REGISTER sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK58289d62
Max-Forwards: 70
From: <sip:9999999999@10.1.1.6>;tag=as399f2e25
To: <sip:9999999999@10.1.1.6>
Call-ID: 0ab72dd24cc63eda7962f1922717ab1d@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="9999999999", realm="asterisk", algorithm=MD5, uri="sip:10.1.1.6", nonce="3315f650", response="e922bb8495c1a8c7aab0b5d518a94172"
Expires: 120
Contact: <sip:9999999999@10.1.1.220:5060>
Content-Length: 0


---
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c:    -- Re-registration for  4444444444@10.1.1.6
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
REGISTER sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK194dfba9
Max-Forwards: 70
From: <sip:4444444444@10.1.1.6>;tag=as0157165c
To: <sip:4444444444@10.1.1.6>
Call-ID: 5853c07806c8b86a4203844e7349692d@10.1.1.220
CSeq: 410 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="4444444444", realm="asterisk", algorithm=MD5, uri="sip:10.1.1.6", nonce="19362624", response="f137d1e552799cd31f58eee509ccd598"
Expires: 120
Contact: <sip:4444444444@10.1.1.220:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
OPTIONS sip:9999999999@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK0aa9e12d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as08779be0
To: <sip:9999999999@10.1.1.220:5060>
Contact: <sip:Unknown@10.1.1.6>
Call-ID: 65f598974298987b175915823aa8e26b@10.1.1.6
CSeq: 102 OPTIONS
User-Agent: TG200V2
Date: Mon, 20 Nov 2017 08:30:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.6:5060 (no NAT)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Looking for 9999999999 in public (domain 10.1.1.220)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK0aa9e12d;received=10.1.1.6;rport=5060
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as08779be0
To: <sip:9999999999@10.1.1.220:5060>;tag=as54796a1e
Call-ID: 65f598974298987b175915823aa8e26b@10.1.1.6
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '65f598974298987b175915823aa8e26b@10.1.1.6' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK58289d62;received=10.1.1.220
From: <sip:9999999999@10.1.1.6>;tag=as399f2e25
To: <sip:9999999999@10.1.1.6>;tag=as30f7a95a
Call-ID: 0ab72dd24cc63eda7962f1922717ab1d@10.1.1.220
CSeq: 411 REGISTER
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:9999999999@10.1.1.220:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:53 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 10.1.1.6 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '0ab72dd24cc63eda7962f1922717ab1d@10.1.1.220' Method: REGISTER
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK194dfba9;received=10.1.1.220
From: <sip:4444444444@10.1.1.6>;tag=as0157165c
To: <sip:4444444444@10.1.1.6>;tag=as2aba5db2
Call-ID: 5853c07806c8b86a4203844e7349692d@10.1.1.220
CSeq: 410 REGISTER
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79dff0de"
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Responding to challenge, registration to domain/host name 10.1.1.6
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
REGISTER sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK19d34f6e
Max-Forwards: 70
From: <sip:4444444444@10.1.1.6>;tag=as0157165c
To: <sip:4444444444@10.1.1.6>
Call-ID: 5853c07806c8b86a4203844e7349692d@10.1.1.220
CSeq: 411 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="4444444444", realm="asterisk", algorithm=MD5, uri="sip:10.1.1.6", nonce="79dff0de", response="e306404643897008eab3cc16d5f6bfcf"
Expires: 120
Contact: <sip:4444444444@10.1.1.220:5060>
Content-Length: 0


---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
OPTIONS sip:4444444444@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK1befa4b9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as2b285754
To: <sip:4444444444@10.1.1.220:5060>
Contact: <sip:Unknown@10.1.1.6>
Call-ID: 79e999532fb73cf5192c3bc90b927a6f@10.1.1.6
CSeq: 102 OPTIONS
User-Agent: TG200V2
Date: Mon, 20 Nov 2017 08:30:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.6:5060 (no NAT)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Looking for 4444444444 in public (domain 10.1.1.220)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK1befa4b9;received=10.1.1.6;rport=5060
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as2b285754
To: <sip:4444444444@10.1.1.220:5060>;tag=as22541f6b
Call-ID: 79e999532fb73cf5192c3bc90b927a6f@10.1.1.6
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '79e999532fb73cf5192c3bc90b927a6f@10.1.1.6' in 32000 ms (Method: OPTIONS)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK19d34f6e;received=10.1.1.220
From: <sip:4444444444@10.1.1.6>;tag=as0157165c
To: <sip:4444444444@10.1.1.6>;tag=as2aba5db2
Call-ID: 5853c07806c8b86a4203844e7349692d@10.1.1.220
CSeq: 411 REGISTER
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:4444444444@10.1.1.220:5060>;expires=120
Date: Mon, 20 Nov 2017 08:30:53 GMT
Content-Length: 0

<------------->
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:30:51] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 10.1.1.6 is 120 sec (Scheduling reregistration in 105 s)
[Nov 20 10:30:51] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '5853c07806c8b86a4203844e7349692d@10.1.1.220' Method: REGISTER
[Nov 20 10:30:58] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '57f107620ebc767e6f27c9df04b40d7d@10.1.1.6' Method: OPTIONS
[Nov 20 10:30:58] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '147950b1012f6d080255f11b1cf0c6bb@10.1.1.6' Method: OPTIONS
[Nov 20 10:31:00] VERBOSE[100135] asterisk.c: Remote UNIX connection
[Nov 20 10:31:00] VERBOSE[100915] asterisk.c: Remote UNIX connection disconnected
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
INVITE sip:748@10.1.1.220;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKfc0bbb6382024cfc
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq:72 INVITE
Contact: <sip:612@10.1.1.15:5060>
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Type: application/sdp
Content-Length: 307

v=0
o=612 1807975390 1807975390 IN IP4 10.1.1.15
s=Session SDP
c=IN IP4 10.1.1.15
t=0 0
m=audio 9002 RTP/AVP 8 4 18 2 0
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=sendrecv
<------------->
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: --- (14 headers 15 lines) ---
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Using INVITE request as basis request - 1B12-1220-46700780A87BC18B73B3-009@SipHost
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found peer '612' for '612' from 10.1.1.15:5060
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKfc0bbb6382024cfc;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>;tag=as1f68a8d2
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq: 72 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58524ca0"
Content-Length: 0


<------------>
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46700780A87BC18B73B3-009@SipHost' in 32000 ms (Method: INVITE)
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
ACK sip:748@10.1.1.220;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKfc0bbb6382024cfc
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>;tag=as1f68a8d2
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq:72 ACK
Max-Forwards:70
Content-Length: 0

<------------->
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: --- (8 headers 0 lines) ---
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
INVITE sip:748@10.1.1.220;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK925d641ae16ac952
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq:73 INVITE
Contact: <sip:612@10.1.1.15:5060>
Expires:90
Max-Forwards:70
Authorization:Digest username="612",realm="asterisk",nonce="58524ca0",uri="sip:748@10.1.1.220;user=phone",response="7cdb4db3e6b188d39c0c194023a3763b",algorithm=MD5
Supported: replaces
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Type: application/sdp
Content-Length: 307

v=0
o=612 1807975390 1807975390 IN IP4 10.1.1.15
s=Session SDP
c=IN IP4 10.1.1.15
t=0 0
m=audio 9002 RTP/AVP 8 4 18 2 0
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: --- (15 headers 15 lines) ---
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Using INVITE request as basis request - 1B12-1220-46700780A87BC18B73B3-009@SipHost
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found peer '612' for '612' from 10.1.1.15:5060
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 4
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 18
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 2
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 0
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format G723 for ID 4
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format G729 for ID 18
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g726|g723|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Peer audio RTP is at port 10.1.1.15:9002
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Looking for 748 in local-stock (domain 10.1.1.220)
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] sip/route.c: sip_route_dump: route/path hop: <sip:612@10.1.1.15:5060>
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK925d641ae16ac952;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq: 73 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:748@10.1.1.220:5060>
Content-Length: 0


<------------>
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] pbx.c: Executing [748@local-stock:1] Dial("SIP/612-00000099", "Sip/3333333/5555555,,D(748),,tT") in new stack
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] chan_sip.c: Audio is at 10098
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK341fd499
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2139282110 2139282110 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10098 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
[Nov 20 10:31:10] VERBOSE[100917][C-00000041] app_dial.c: Called Sip/3333333/5555555
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK341fd499;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as4458f8de
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7212ccf3"
Content-Length: 0

<------------->
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK341fd499
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as4458f8de
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Audio is at 10098
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:10] VERBOSE[100192][C-00000041] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK28a59f79
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="3333333", realm="asterisk", algorithm=MD5, uri="sip:5555555@12.34.56.78", nonce="7212ccf3", response="6b143a6238569a8a4ae3dda9a23c4178"
Date: Mon, 20 Nov 2017 08:31:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2139282110 2139282111 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10098 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK28a59f79;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Length: 0

<------------->
[Nov 20 10:31:10] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK28a59f79;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 994357930 994357930 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 19934 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
<------------->
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] sip/route.c: sip_route_dump: route/path hop: <sip:5555555@12.34.56.78>
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Peer audio RTP is at port 12.34.56.78:19934
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] app_dial.c: SIP/3333333-0000009a is making progress passing it to SIP/612-00000099
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Audio is at 10010
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK925d641ae16ac952;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>;tag=as71fee1f9
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq: 73 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:748@10.1.1.220:5060>
Content-Type: application/sdp
Content-Length: 203

v=0
o=root 1799932873 1799932873 IN IP4 10.1.1.220
s=Asterisk PBX 13.6.0
c=IN IP4 10.1.1.220
t=0 0
m=audio 10010 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv

<------------>
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] res_rtp_asterisk.c: 0x82dd24000 -- Probation passed - setting RTP source address to 12.34.56.78:19934
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] res_rtp_asterisk.c: 0x82dc98000 -- Probation passed - setting RTP source address to 10.1.1.15:9002
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK28a59f79;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 994357930 994357931 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 19934 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
<------------->
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Peer audio RTP is at port 12.34.56.78:19934
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] sip/route.c: sip_route_dump: route/path hop: <sip:5555555@12.34.56.78>
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK2873fa2d
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] app_dial.c: SIP/3333333-0000009a answered SIP/612-00000099
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Audio is at 10098
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK79b623f3
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 104 INVITE
User-Agent: Asterisk PBX 13.6.0
Access-URL: <,tT>;mode=active
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2139282110 2139282112 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10098 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] app_dial.c: Sending DTMF '748' to the called party.
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] res_rtp_asterisk.c: 0x82dd24000 -- Probation passed - setting RTP source address to 12.34.56.78:19934
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK79b623f3;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 104 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Length: 0

<------------->
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK79b623f3;received=87.65.43.21
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 104 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 994357930 994357932 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 19934 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
<------------->
[Nov 20 10:31:11] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Peer audio RTP is at port 12.34.56.78:19934
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:11] VERBOSE[100192][C-00000041] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK701f0fec
Max-Forwards: 70
From: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
To: <sip:5555555@12.34.56.78>;tag=as68f0b575
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 104 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:11] VERBOSE[100917][C-00000041] res_rtp_asterisk.c: 0x82dd24000 -- Probation passed - setting RTP source address to 12.34.56.78:19934
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] chan_sip.c: Audio is at 10010
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK925d641ae16ac952;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>;tag=as71fee1f9
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq: 73 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:748@10.1.1.220:5060>
Content-Type: application/sdp
Content-Length: 203

v=0
o=root 1799932873 1799932873 IN IP4 10.1.1.220
s=Asterisk PBX 13.6.0
c=IN IP4 10.1.1.220
t=0 0
m=audio 10010 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv

<------------>
[Nov 20 10:31:12] VERBOSE[100919][C-00000041] bridge_channel.c: Channel SIP/3333333-0000009a joined 'simple_bridge' basic-bridge <e61cf0c9-b52a-4430-b235-d1535bccdcf1>
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] bridge_channel.c: Channel SIP/612-00000099 joined 'simple_bridge' basic-bridge <e61cf0c9-b52a-4430-b235-d1535bccdcf1>
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] bridge.c: Bridge e61cf0c9-b52a-4430-b235-d1535bccdcf1: switching from simple_bridge technology to native_rtp
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] bridge_native_rtp.c: Locally RTP bridged 'SIP/612-00000099' and 'SIP/3333333-0000009a' in stack
[Nov 20 10:31:12] VERBOSE[100917][C-00000041] bridge_native_rtp.c: Locally RTP bridged 'SIP/612-00000099' and 'SIP/3333333-0000009a' in stack
[Nov 20 10:31:13] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
ACK sip:748@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK8745d404aeeeb673
From: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
To: <sip:748@10.1.1.220;user=phone>;tag=as71fee1f9
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq:73 ACK
Max-Forwards:70
Authorization:Digest username="612",realm="asterisk",nonce="58524ca0",uri="sip:748@10.1.1.220;user=phone",response="7cdb4db3e6b188d39c0c194023a3763b",algorithm=MD5
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:31:13] VERBOSE[100192] chan_sip.c: --- (10 headers 0 lines) ---
[Nov 20 10:31:15] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1B12-1220-466848125E5C7B2412BD-001@SipHost' Method: REGISTER
[Nov 20 10:31:15] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1B12-1220-466848122EBAD76C7494-002@SipHost' Method: REGISTER
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
OPTIONS sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK7ebedcc1
Max-Forwards: 70
From: "Unknown" <sip:4444444444@10.1.1.220>;tag=as1019a4ee
To: <sip:10.1.1.6>
Contact: <sip:4444444444@10.1.1.220:5060>
Call-ID: 7d9232002fac01836b00affe7391ff57@10.1.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK7ebedcc1;received=10.1.1.220
From: "Unknown" <sip:4444444444@10.1.1.220>;tag=as1019a4ee
To: <sip:10.1.1.6>;tag=as3606e20d
Call-ID: 7d9232002fac01836b00affe7391ff57@10.1.1.220:5060
CSeq: 102 OPTIONS
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '7d9232002fac01836b00affe7391ff57@10.1.1.220:5060' Method: OPTIONS
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
OPTIONS sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK4e739d35
Max-Forwards: 70
From: "Unknown" <sip:9999999999@10.1.1.220>;tag=as3bfbdf51
To: <sip:10.1.1.6>
Contact: <sip:9999999999@10.1.1.220:5060>
Call-ID: 0670291c1c7a3cbf1638fe680b60083d@10.1.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK4e739d35;received=10.1.1.220
From: "Unknown" <sip:9999999999@10.1.1.220>;tag=as3bfbdf51
To: <sip:10.1.1.6>;tag=as23fa1a2b
Call-ID: 0670291c1c7a3cbf1638fe680b60083d@10.1.1.220:5060
CSeq: 102 OPTIONS
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '0670291c1c7a3cbf1638fe680b60083d@10.1.1.220:5060' Method: OPTIONS
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK1e56dd7b
Max-Forwards: 70
From: "Unknown" <sip:8888888@87.65.43.21>;tag=as461c11b1
To: <sip:12.34.56.78>
Contact: <sip:8888888@87.65.43.21:5060>
Call-ID: 27d0499b3c297f0d057715a54bc3565e@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK02e1fada
Max-Forwards: 70
From: "Unknown" <sip:3333333@87.65.43.21>;tag=as2214650f
To: <sip:12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 7bf132e754a7f2093085aa2916e0e3de@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK5f2cf0a6
Max-Forwards: 70
From: "Unknown" <sip:6666666@87.65.43.21>;tag=as48666d72
To: <sip:12.34.56.78>
Contact: <sip:6666666@87.65.43.21:5060>
Call-ID: 2e32290e454d38334de50cbb504483c4@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK6ec17d3b
Max-Forwards: 70
From: "Unknown" <sip:7777777@87.65.43.21>;tag=as0bfa8e8f
To: <sip:12.34.56.78>
Contact: <sip:7777777@87.65.43.21:5060>
Call-ID: 49a4df35377381734ced9cb65d986a90@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK1e56dd7b;received=87.65.43.21
From: "Unknown" <sip:8888888@87.65.43.21>;tag=as461c11b1
To: <sip:12.34.56.78>;tag=as3b9f76e7
Call-ID: 27d0499b3c297f0d057715a54bc3565e@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '27d0499b3c297f0d057715a54bc3565e@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK02e1fada;received=87.65.43.21
From: "Unknown" <sip:3333333@87.65.43.21>;tag=as2214650f
To: <sip:12.34.56.78>;tag=as3fe7cc95
Call-ID: 7bf132e754a7f2093085aa2916e0e3de@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '7bf132e754a7f2093085aa2916e0e3de@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK5f2cf0a6;received=87.65.43.21
From: "Unknown" <sip:6666666@87.65.43.21>;tag=as48666d72
To: <sip:12.34.56.78>;tag=as09a6d6e8
Call-ID: 2e32290e454d38334de50cbb504483c4@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '2e32290e454d38334de50cbb504483c4@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK6ec17d3b;received=87.65.43.21
From: "Unknown" <sip:7777777@87.65.43.21>;tag=as0bfa8e8f
To: <sip:12.34.56.78>;tag=as799eb091
Call-ID: 49a4df35377381734ced9cb65d986a90@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '49a4df35377381734ced9cb65d986a90@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:31:22] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '5fce3cd43bf064b2333fd4f307b52b3f@12.34.56.78' Method: OPTIONS
[Nov 20 10:31:22] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '6e3565174e1679496ea3ccdb0432e37f@12.34.56.78' Method: OPTIONS
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '14a823a1033d53420110c2a460e36da1@12.34.56.78' Method: OPTIONS
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '11c7aa675bcea45e71260fd82ce40c60@12.34.56.78' Method: OPTIONS
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.164:5060 --->
INVITE sip:745@010.1.1.220 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK750340611;rport
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 20 INVITE
Contact: <sip:664@10.1.1.164:5060>
Max-Forwards: 70
User-Agent: Grandstream GXP1620 1.0.2.27
Privacy: none
P-Preferred-Identity: <sip:664@010.1.1.220>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 328

v=0
o=664 8000 8000 IN IP4 10.1.1.164
s=SIP Call
c=IN IP4 10.1.1.164
t=0 0
m=audio 5004 RTP/AVP 0 8 18 9 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: --- (16 headers 16 lines) ---
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.164:5060 (no NAT)
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Sending to 10.1.1.164:5060 (no NAT)
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Using INVITE request as basis request - 470514923-5060-2@BA.B.B.BGE
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found peer '664' for '664' from 10.1.1.164:5060
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 0
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 18
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 9
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 2
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format G729 for ID 18
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format G722 for ID 9
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format G726-32 for ID 2
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g726|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Peer audio RTP is at port 10.1.1.164:5004
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Looking for 745 in local-emarket (domain 010.1.1.220)
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] sip/route.c: sip_route_dump: route/path hop: <sip:664@10.1.1.164:5060>
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.164:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK750340611;received=10.1.1.164;rport=5060
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 20 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:745@10.1.1.220:5060>
Content-Length: 0


<------------>
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] pbx.c: Executing [745@local-emarket:1] Dial("SIP/664-0000009b", "Sip/3333333/5555555,,D(745),,tT") in new stack
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] chan_sip.c: Audio is at 10070
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK6f98dc79
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1676792322 1676792322 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10070 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
[Nov 20 10:31:23] VERBOSE[100920][C-00000042] app_dial.c: Called Sip/3333333/5555555
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK6f98dc79;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as66d6945e
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="31b6f819"
Content-Length: 0

<------------->
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK6f98dc79
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as66d6945e
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Audio is at 10070
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:23] VERBOSE[100192][C-00000042] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21edf7f6
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="3333333", realm="asterisk", algorithm=MD5, uri="sip:5555555@12.34.56.78", nonce="31b6f819", response="759bc5213fd8c8f74e989382446673be"
Date: Mon, 20 Nov 2017 08:31:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1676792322 1676792323 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10070 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telep[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21edf7f6;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Length: 0

<------------->
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '65f598974298987b175915823aa8e26b@10.1.1.6' Method: OPTIONS
[Nov 20 10:31:23] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '79e999532fb73cf5192c3bc90b927a6f@10.1.1.6' Method: OPTIONS
[Nov 20 10:31:24] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21edf7f6;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 713484694 713484694 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 15332 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
<------------->
[Nov 20 10:31:24] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] sip/route.c: sip_route_dump: route/path hop: <sip:5555555@12.34.56.78>
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:24] VERBOSE[100192][C-00000042] chan_sip.c: Peer audio RTP is at port 12.34.56.78:15332
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] app_dial.c: SIP/3333333-0000009c is making progress passing it to SIP/664-0000009b
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] chan_sip.c: Audio is at 10038
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.164:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK750340611;received=10.1.1.164;rport=5060
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>;tag=as23c1b8c1
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 20 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:745@10.1.1.220:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 259

v=0
o=root 1793092479 1793092479 IN IP4 10.1.1.220
s=Asterisk PBX 13.6.0
c=IN IP4 10.1.1.220
t=0 0
m=audio 10038 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] res_rtp_asterisk.c: 0x82de5f000 -- Probation passed - setting RTP source address to 12.34.56.78:15332
[Nov 20 10:31:24] VERBOSE[100920][C-00000042] res_rtp_asterisk.c: 0x82e087000 -- Probation passed - setting RTP source address to 10.1.1.164:5004
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK21edf7f6;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 103 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 713484694 713484695 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 15332 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
<------------->
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] app_dial.c: SIP/3333333-0000009c requested media update control 26, passing it to SIP/664-0000009b
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Peer audio RTP is at port 12.34.56.78:15332
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] sip/route.c: sip_route_dump: route/path hop: <sip:5555555@12.34.56.78>
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK4ff76813
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] app_dial.c: SIP/3333333-0000009c answered SIP/664-0000009b
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: Audio is at 10070
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
INVITE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK4056c8a8
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 104 INVITE
User-Agent: Asterisk PBX 13.6.0
Access-URL: <,tT>;mode=active
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1676792322 1676792324 IN IP4 87.65.43.21
s=Asterisk PBX 13.6.0
c=IN IP4 87.65.43.21
t=0 0
m=audio 10070 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] app_dial.c: Sending DTMF '745' to the called party.
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] res_rtp_asterisk.c: 0x82de5f000 -- Probation passed - setting RTP source address to 12.34.56.78:15332
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK4056c8a8;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 104 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Length: 0

<------------->
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK4056c8a8;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 104 INVITE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:5555555@12.34.56.78>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 713484694 713484696 IN IP4 12.34.56.78
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 12.34.56.78
t=0 0
m=audio 15332 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
<------------->
[Nov 20 10:31:25] VERBOSE[100192] chan_sip.c: --- (12 headers 12 lines) ---
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 8
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found RTP audio format 101
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Peer audio RTP is at port 12.34.56.78:15332
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:25] VERBOSE[100192][C-00000042] chan_sip.c: Transmitting (no NAT) to 12.34.56.78:5060:
ACK sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK33a2778e
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 104 ACK
User-Agent: Asterisk PBX 13.6.0
Content-Length: 0


---
[Nov 20 10:31:25] VERBOSE[100920][C-00000042] res_rtp_asterisk.c: 0x82de5f000 -- Probation passed - setting RTP source address to 12.34.56.78:15332
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] chan_sip.c: Audio is at 10038
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec alaw to SDP
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] chan_sip.c: Adding codec ulaw to SDP
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK750340611;received=10.1.1.164;rport=5060
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>;tag=as23c1b8c1
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 20 INVITE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:745@10.1.1.220:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 259

v=0
o=root 1793092479 1793092479 IN IP4 10.1.1.220
s=Asterisk PBX 13.6.0
c=IN IP4 10.1.1.220
t=0 0
m=audio 10038 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
[Nov 20 10:31:26] VERBOSE[100921][C-00000042] bridge_channel.c: Channel SIP/3333333-0000009c joined 'simple_bridge' basic-bridge <6d8ec6fc-5065-4035-83ca-c5214b916158>
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] bridge_channel.c: Channel SIP/664-0000009b joined 'simple_bridge' basic-bridge <6d8ec6fc-5065-4035-83ca-c5214b916158>
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] bridge.c: Bridge 6d8ec6fc-5065-4035-83ca-c5214b916158: switching from simple_bridge technology to native_rtp
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] bridge_native_rtp.c: Locally RTP bridged 'SIP/664-0000009b' and 'SIP/3333333-0000009c' in stack
[Nov 20 10:31:26] VERBOSE[100920][C-00000042] bridge_native_rtp.c: Locally RTP bridged 'SIP/664-0000009b' and 'SIP/3333333-0000009c' in stack
[Nov 20 10:31:26] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.164:5060 --->
ACK sip:745@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK1394575353;rport
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>;tag=as23c1b8c1
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 20 ACK
Contact: <sip:664@10.1.1.164:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1620 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[Nov 20 10:31:26] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:30] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.23.34.45:5060:
OPTIONS sip:12.23.34.45 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK0b39ff5b
Max-Forwards: 70
From: "Unknown" <sip:380894201108@87.65.43.21>;tag=as56764720
To: <sip:12.23.34.45>
Contact: <sip:380894201108@87.65.43.21:5060>
Call-ID: 645e1f1158cd4874064464d716f03153@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:31:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:31:30] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.23.34.45:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK0b39ff5b
To: <sip:12.23.34.45>;tag=mvfnmp3huht46opq
From: Unknown <sip:380894201108@87.65.43.21>;tag=as56764720
Call-ID: 645e1f1158cd4874064464d716f03153@87.65.43.21:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Server: Sippy
Supported: replaces
Content-Length: 0

<------------->
[Nov 20 10:31:30] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:30] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '645e1f1158cd4874064464d716f03153@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:31:38] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.164:5060 --->
BYE sip:745@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK336576392;rport
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>;tag=as23c1b8c1
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 21 BYE
Contact: <sip:664@10.1.1.164:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1620 1.0.2.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[Nov 20 10:31:38] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:38] VERBOSE[100192][C-00000042] chan_sip.c: Sending to 10.1.1.164:5060 (no NAT)
[Nov 20 10:31:38] VERBOSE[100192][C-00000042] chan_sip.c: Scheduling destruction of SIP dialog '470514923-5060-2@BA.B.B.BGE' in 32000 ms (Method: BYE)
[Nov 20 10:31:38] VERBOSE[100192][C-00000042] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.164:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.164:5060;branch=z9hG4bK336576392;received=10.1.1.164;rport=5060
From: <sip:664@010.1.1.220>;tag=870807617
To: <sip:745@010.1.1.220>;tag=as23c1b8c1
Call-ID: 470514923-5060-2@BA.B.B.BGE
CSeq: 21 BYE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Nov 20 10:31:38] VERBOSE[100920][C-00000042] bridge_channel.c: Channel SIP/664-0000009b left 'native_rtp' basic-bridge <6d8ec6fc-5065-4035-83ca-c5214b916158>
[Nov 20 10:31:38] VERBOSE[100920][C-00000042] pbx.c: Spawn extension (local-emarket, 745, 1) exited non-zero on 'SIP/664-0000009b'
[Nov 20 10:31:38] VERBOSE[100921][C-00000042] bridge_channel.c: Channel SIP/3333333-0000009c left 'native_rtp' basic-bridge <6d8ec6fc-5065-4035-83ca-c5214b916158>
[Nov 20 10:31:38] VERBOSE[100921][C-00000042] chan_sip.c: Scheduling destruction of SIP dialog '68866b43442bfc8b1979196070955ff1@87.65.43.21' in 6400 ms (Method: INVITE)
[Nov 20 10:31:38] VERBOSE[100921][C-00000042] chan_sip.c: set_destination: Parsing <sip:5555555@12.34.56.78> for address/port to send to
[Nov 20 10:31:38] VERBOSE[100921][C-00000042] chan_sip.c: set_destination: set destination to 12.34.56.78:5060
[Nov 20 10:31:38] VERBOSE[100921][C-00000042] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
BYE sip:5555555@12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK7b3220a2
Max-Forwards: 70
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 105 BYE
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="3333333", realm="asterisk", algorithm=MD5, uri="sip:5555555@12.34.56.78", nonce="31b6f819", response="d86ed27ecb3e405b0e95778dbd20cff4"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov 20 10:31:38] VERBOSE[100920][C-00000042] pbx.c: Executing [h@local-emarket:1] ExecIf("SIP/664-0000009b", "0?Set(ODBC_asteriskHGOU()=745)") in new stack
[Nov 20 10:31:38] VERBOSE[100920][C-00000042] pbx.c: Executing [h@local-emarket:2] Set("SIP/664-0000009b", "timesql="2017-11-20T10:31:38"") in new stack
[Nov 20 10:31:38] VERBOSE[100920][C-00000042] pbx.c: Executing [h@local-emarket:3] ExecIf("SIP/664-0000009b", "0?Set(ODBC_fixDuration()=,,"2017-11-20T10:31:38")") in new stack
[Nov 20 10:31:38] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK7b3220a2;received=87.65.43.21
From: <sip:3333333@87.65.43.21>;tag=as79d90720
To: <sip:5555555@12.34.56.78>;tag=as224669dc
Call-ID: 68866b43442bfc8b1979196070955ff1@87.65.43.21
CSeq: 105 BYE
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:31:38] VERBOSE[100192] chan_sip.c: --- (10 headers 0 lines) ---
[Nov 20 10:31:38] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '68866b43442bfc8b1979196070955ff1@87.65.43.21' Method: INVITE
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:6666666@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK569603d1;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as09c28c0a
To: <sip:6666666@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 59edbacc48c295c82bc593aa74e8dd5a@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:31:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: Looking for 6666666 in public (domain 87.65.43.21)
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK569603d1;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as09c28c0a
To: <sip:6666666@87.65.43.21:5060>;tag=as67258e50
Call-ID: 59edbacc48c295c82bc593aa74e8dd5a@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:31:46] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '59edbacc48c295c82bc593aa74e8dd5a@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKa0dc4fcba21b03cc
From: "611" <sip:611@10.1.1.220>;tag=2c9d2faa-684812
To: "611" <sip:611@10.1.1.220>
Call-ID: 1B12-1220-46684812A5C42D328A7F-001@SipHost
CSeq:1 REGISTER
Contact: <sip:611@10.1.1.15:5060>
Expires:600
Max-Forwards:70
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bKa0dc4fcba21b03cc;received=10.1.1.15
From: "611" <sip:611@10.1.1.220>;tag=2c9d2faa-684812
To: "611" <sip:611@10.1.1.220>;tag=as6f7274df
Call-ID: 1B12-1220-46684812A5C42D328A7F-001@SipHost
CSeq: 1 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a923bd6"
Content-Length: 0


<------------>
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46684812A5C42D328A7F-001@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK51800421f60322a5
From: "612" <sip:612@10.1.1.220>;tag=8cffa7a7-684812
To: "612" <sip:612@10.1.1.220>
Call-ID: 1B12-1220-46684812F94D7B15C13F-002@SipHost
CSeq:1 REGISTER
Contact: <sip:612@10.1.1.15:5060>
Expires:600
Max-Forwards:70
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK51800421f60322a5;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=8cffa7a7-684812
To: "612" <sip:612@10.1.1.220>;tag=as50db65aa
Call-ID: 1B12-1220-46684812F94D7B15C13F-002@SipHost
CSeq: 1 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="326e42a9"
Content-Length: 0


<------------>
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46684812F94D7B15C13F-002@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK0dc453f76de2ad31
From: "611" <sip:611@10.1.1.220>;tag=2c9d2faa-684812
To: "611" <sip:611@10.1.1.220>
Call-ID: 1B12-1220-46684812A5C42D328A7F-001@SipHost
CSeq:2 REGISTER
Contact: <sip:611@10.1.1.15:5060>
Expires:600
Max-Forwards:70
Authorization:Digest username="611",realm="asterisk",nonce="2a923bd6",uri="sip:10.1.1.220:5060",response="b04c42652e032bea37d26913c410e24f",algorithm=MD5
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK0dc453f76de2ad31;received=10.1.1.15
From: "611" <sip:611@10.1.1.220>;tag=2c9d2faa-684812
To: "611" <sip:611@10.1.1.220>;tag=as6f7274df
Call-ID: 1B12-1220-46684812A5C42D328A7F-001@SipHost
CSeq: 2 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:611@10.1.1.15:5060>;expires=600
Date: Mon, 20 Nov 2017 08:31:47 GMT
Content-Length: 0


<------------>
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46684812A5C42D328A7F-001@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
REGISTER sip:10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK3474360040bfdadc
From: "612" <sip:612@10.1.1.220>;tag=8cffa7a7-684812
To: "612" <sip:612@10.1.1.220>
Call-ID: 1B12-1220-46684812F94D7B15C13F-002@SipHost
CSeq:2 REGISTER
Contact: <sip:612@10.1.1.15:5060>
Expires:600
Max-Forwards:70
Authorization:Digest username="612",realm="asterisk",nonce="326e42a9",uri="sip:10.1.1.220:5060",response="472b0cc915ef420dc86043db053836a6",algorithm=MD5
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.15:5060 (no NAT)
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.15:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK3474360040bfdadc;received=10.1.1.15
From: "612" <sip:612@10.1.1.220>;tag=8cffa7a7-684812
To: "612" <sip:612@10.1.1.220>;tag=as50db65aa
Call-ID: 1B12-1220-46684812F94D7B15C13F-002@SipHost
CSeq: 2 REGISTER
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 600
Contact: <sip:612@10.1.1.15:5060>;expires=600
Date: Mon, 20 Nov 2017 08:31:47 GMT
Content-Length: 0


<------------>
[Nov 20 10:31:47] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46684812F94D7B15C13F-002@SipHost' in 32000 ms (Method: REGISTER)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:3333333@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK084d60b1;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as42518bfa
To: <sip:3333333@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 698dcb1b6985536730a4c2ff4082404f@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:31:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Looking for 3333333 in public (domain 87.65.43.21)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK084d60b1;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as42518bfa
To: <sip:3333333@87.65.43.21:5060>;tag=as43a0dec0
Call-ID: 698dcb1b6985536730a4c2ff4082404f@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '698dcb1b6985536730a4c2ff4082404f@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:8888888@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK65f4ca34;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as5a72ceb1
To: <sip:8888888@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 595f135427a096de49cbf2290dc43170@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:31:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Looking for 8888888 in public (domain 87.65.43.21)
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK65f4ca34;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as5a72ceb1
To: <sip:8888888@87.65.43.21:5060>;tag=as581a3d60
Call-ID: 595f135427a096de49cbf2290dc43170@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:31:50] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '595f135427a096de49cbf2290dc43170@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
OPTIONS sip:7777777@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK4eafb694;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as1a33460b
To: <sip:7777777@87.65.43.21:5060>
Contact: <sip:Unknown@12.34.56.78>
Call-ID: 3066a891389cd78809a6446d10d43d6c@12.34.56.78
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.6.2.16.1)
Date: Mon, 20 Nov 2017 08:31:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: Looking for 7777777 in public (domain 87.65.43.21)
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK4eafb694;received=12.34.56.78;rport=5060
From: "Unknown" <sip:Unknown@12.34.56.78>;tag=as1a33460b
To: <sip:7777777@87.65.43.21:5060>;tag=as43a3b5bd
Call-ID: 3066a891389cd78809a6446d10d43d6c@12.34.56.78
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:31:51] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '3066a891389cd78809a6446d10d43d6c@12.34.56.78' in 32000 ms (Method: OPTIONS)
[Nov 20 10:32:00] VERBOSE[100135] asterisk.c: Remote UNIX connection
[Nov 20 10:32:00] VERBOSE[100922] asterisk.c: Remote UNIX connection disconnected
[Nov 20 10:32:07] NOTICE[100192] chan_sip.c:    -- Re-registration for  380894201108@12.23.34.45
[Nov 20 10:32:07] VERBOSE[100192] chan_sip.c: REGISTER 12 headers, 0 lines
[Nov 20 10:32:07] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.23.34.45:5060:
REGISTER sip:12.23.34.45 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK545ed130
Max-Forwards: 70
From: <sip:380894201108@12.23.34.45>;tag=as25614221
To: <sip:380894201108@12.23.34.45>
Call-ID: 2e69d0bf46a8483004843fff4fa909e4@10.1.1.220
CSeq: 188 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.6.0
Authorization: Digest username="380894201108", realm="sip-1.pb", algorithm=MD5, uri="sip:12.23.34.45", nonce="1511166444:dab62deb31443f1fa602a54abc81b3eb", response="e4c392f2c9a7e85ca9776c730f770a9a"
Expires: 120
Contact: <sip:380894201108@87.65.43.21:5060>
Content-Length: 0


---
[Nov 20 10:32:07] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.23.34.45:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK545ed130
Contact: <sip:380894201108@87.65.43.21:54303>;expires=1356
Contact: <sip:380894201108@87.65.43.21:5060>;expires=300
To: <sip:380894201108@12.23.34.45>;tag=48cf4f35
From: <sip:380894201108@12.23.34.45>;tag=as25614221
Call-ID: 2e69d0bf46a8483004843fff4fa909e4@10.1.1.220
CSeq: 188 REGISTER
Date: Mon, 20 Nov 2017 08:32:09 GMT
PortaBilling: available-funds:1197.60945 currency:UAH
Content-Length: 0

<------------->
[Nov 20 10:32:07] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:32:07] NOTICE[100192] chan_sip.c: Outbound Registration: Expiry for 12.23.34.45 is 300 sec (Scheduling reregistration in 285 s)
[Nov 20 10:32:07] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '2e69d0bf46a8483004843fff4fa909e4@10.1.1.220' Method: REGISTER
[Nov 20 10:32:08] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
BYE sip:3333333@87.65.43.21:5060 SIP/2.0
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0126f84f;rport
Max-Forwards: 70
From: <sip:5555555@12.34.56.78>;tag=as68f0b575
To: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 102 BYE
User-Agent: FPBX-2.9.0(1.6.2.16.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
[Nov 20 10:32:08] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:32:08] VERBOSE[100192][C-00000041] chan_sip.c: Sending to 12.34.56.78:5060 (no NAT)
[Nov 20 10:32:08] VERBOSE[100192][C-00000041] chan_sip.c: Scheduling destruction of SIP dialog '1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21' in 6400 ms (Method: BYE)
[Nov 20 10:32:08] VERBOSE[100192][C-00000041] chan_sip.c: 
<--- Transmitting (no NAT) to 12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.34.56.78:5060;branch=z9hG4bK0126f84f;received=12.34.56.78;rport=5060
From: <sip:5555555@12.34.56.78>;tag=as68f0b575
To: "612" <sip:3333333@87.65.43.21>;tag=as24a5a2e4
Call-ID: 1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21
CSeq: 102 BYE
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Nov 20 10:32:08] VERBOSE[100919][C-00000041] bridge_channel.c: Channel SIP/3333333-0000009a left 'native_rtp' basic-bridge <e61cf0c9-b52a-4430-b235-d1535bccdcf1>
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] bridge_channel.c: Channel SIP/612-00000099 left 'native_rtp' basic-bridge <e61cf0c9-b52a-4430-b235-d1535bccdcf1>
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] pbx.c: Spawn extension (local-stock, 748, 1) exited non-zero on 'SIP/612-00000099'
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] chan_sip.c: Scheduling destruction of SIP dialog '1B12-1220-46700780A87BC18B73B3-009@SipHost' in 32000 ms (Method: ACK)
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] chan_sip.c: set_destination: Parsing <sip:612@10.1.1.15:5060> for address/port to send to
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] chan_sip.c: set_destination: set destination to 10.1.1.15:5060
[Nov 20 10:32:08] VERBOSE[100917][C-00000041] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.15:5060:
BYE sip:612@10.1.1.15:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK1c5cd2e3
Max-Forwards: 70
From: <sip:748@10.1.1.220;user=phone>;tag=as71fee1f9
To: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.6.0
Proxy-Authorization: Digest username="612", realm="asterisk", algorithm=MD5, uri="sip:10.1.1.220", nonce="58524ca0", response="91c69181b3181f36412c2506cb7b6d42"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov 20 10:32:08] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK1c5cd2e3
From: <sip:748@10.1.1.220;user=phone>;tag=as71fee1f9
To: "612" <sip:612@10.1.1.220>;tag=df385fc1-700780
Call-ID: 1B12-1220-46700780A87BC18B73B3-009@SipHost
CSeq:102 BYE
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Length: 0

<------------->
[Nov 20 10:32:08] VERBOSE[100192] chan_sip.c: --- (8 headers 0 lines) ---
[Nov 20 10:32:08] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1B12-1220-46700780A87BC18B73B3-009@SipHost' Method: ACK
[Nov 20 10:32:10] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '470514923-5060-2@BA.B.B.BGE' Method: BYE
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
OPTIONS sip:9999999999@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK6fafcc9e;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as5d17cf3e
To: <sip:9999999999@10.1.1.220:5060>
Contact: <sip:Unknown@10.1.1.6>
Call-ID: 3f24924f45e8c5360c961b4f00b6f2a8@10.1.1.6
CSeq: 102 OPTIONS
User-Agent: TG200V2
Date: Mon, 20 Nov 2017 08:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.6:5060 (no NAT)
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Looking for 9999999999 in public (domain 10.1.1.220)
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK6fafcc9e;received=10.1.1.6;rport=5060
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as5d17cf3e
To: <sip:9999999999@10.1.1.220:5060>;tag=as332a1788
Call-ID: 3f24924f45e8c5360c961b4f00b6f2a8@10.1.1.6
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '3f24924f45e8c5360c961b4f00b6f2a8@10.1.1.6' in 32000 ms (Method: OPTIONS)
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
OPTIONS sip:4444444444@10.1.1.220:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK1fd9ec27;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as558f74ee
To: <sip:4444444444@10.1.1.220:5060>
Contact: <sip:Unknown@10.1.1.6>
Call-ID: 0773265f0c2733dc3d08a9cf3c733e9a@10.1.1.6
CSeq: 102 OPTIONS
User-Agent: TG200V2
Date: Mon, 20 Nov 2017 08:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: --- (13 headers 0 lines) ---
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Sending to 10.1.1.6:5060 (no NAT)
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Looking for 4444444444 in public (domain 10.1.1.220)
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.6:5060;branch=z9hG4bK1fd9ec27;received=10.1.1.6;rport=5060
From: "Unknown" <sip:Unknown@10.1.1.6>;tag=as558f74ee
To: <sip:4444444444@10.1.1.220:5060>;tag=as26bd72f7
Call-ID: 0773265f0c2733dc3d08a9cf3c733e9a@10.1.1.6
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
[Nov 20 10:32:11] VERBOSE[100192] chan_sip.c: Scheduling destruction of SIP dialog '0773265f0c2733dc3d08a9cf3c733e9a@10.1.1.6' in 32000 ms (Method: OPTIONS)
[Nov 20 10:32:14] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1c5c3e0b00940b2068b667675e09f5e1@87.65.43.21' Method: BYE
[Nov 20 10:32:18] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '59edbacc48c295c82bc593aa74e8dd5a@12.34.56.78' Method: OPTIONS
[Nov 20 10:32:19] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1B12-1220-46684812A5C42D328A7F-001@SipHost' Method: REGISTER
[Nov 20 10:32:19] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '1B12-1220-46684812F94D7B15C13F-002@SipHost' Method: REGISTER
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
OPTIONS sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK32453d9b
Max-Forwards: 70
From: "Unknown" <sip:4444444444@10.1.1.220>;tag=as5b42acdb
To: <sip:10.1.1.6>
Contact: <sip:4444444444@10.1.1.220:5060>
Call-ID: 4272fb784c45695a2751fac15a868743@10.1.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK32453d9b;received=10.1.1.220
From: "Unknown" <sip:4444444444@10.1.1.220>;tag=as5b42acdb
To: <sip:10.1.1.6>;tag=as17fc341a
Call-ID: 4272fb784c45695a2751fac15a868743@10.1.1.220:5060
CSeq: 102 OPTIONS
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '4272fb784c45695a2751fac15a868743@10.1.1.220:5060' Method: OPTIONS
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.6:5060:
OPTIONS sip:10.1.1.6 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK04ccb74d
Max-Forwards: 70
From: "Unknown" <sip:9999999999@10.1.1.220>;tag=as7b2a1773
To: <sip:10.1.1.6>
Contact: <sip:9999999999@10.1.1.220:5060>
Call-ID: 5befbccc32c15fe149249d753991755e@10.1.1.220:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.6:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.220:5060;branch=z9hG4bK04ccb74d;received=10.1.1.220
From: "Unknown" <sip:9999999999@10.1.1.220>;tag=as7b2a1773
To: <sip:10.1.1.6>;tag=as7bb29e04
Call-ID: 5befbccc32c15fe149249d753991755e@10.1.1.220:5060
CSeq: 102 OPTIONS
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (11 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '5befbccc32c15fe149249d753991755e@10.1.1.220:5060' Method: OPTIONS
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK50afcc22
Max-Forwards: 70
From: "Unknown" <sip:8888888@87.65.43.21>;tag=as55355a1b
To: <sip:12.34.56.78>
Contact: <sip:8888888@87.65.43.21:5060>
Call-ID: 147c557364877a933ef058a75cb4603d@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK701f69da
Max-Forwards: 70
From: "Unknown" <sip:3333333@87.65.43.21>;tag=as5ef12ec3
To: <sip:12.34.56.78>
Contact: <sip:3333333@87.65.43.21:5060>
Call-ID: 3ce1ea71543d92630ae334c43ce5f80f@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK74408c15
Max-Forwards: 70
From: "Unknown" <sip:6666666@87.65.43.21>;tag=as32f65233
To: <sip:12.34.56.78>
Contact: <sip:6666666@87.65.43.21:5060>
Call-ID: 295331a045b6fdd71df4d201567ab40c@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Reliably Transmitting (no NAT) to 12.34.56.78:5060:
OPTIONS sip:12.34.56.78 SIP/2.0
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK409385ca
Max-Forwards: 70
From: "Unknown" <sip:7777777@87.65.43.21>;tag=as348c548a
To: <sip:12.34.56.78>
Contact: <sip:7777777@87.65.43.21:5060>
Call-ID: 4e2808c70ce9272f5e71383a23bae10b@87.65.43.21:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.6.0
Date: Mon, 20 Nov 2017 08:32:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK50afcc22;received=87.65.43.21
From: "Unknown" <sip:8888888@87.65.43.21>;tag=as55355a1b
To: <sip:12.34.56.78>;tag=as7463acd5
Call-ID: 147c557364877a933ef058a75cb4603d@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '147c557364877a933ef058a75cb4603d@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK701f69da;received=87.65.43.21
From: "Unknown" <sip:3333333@87.65.43.21>;tag=as5ef12ec3
To: <sip:12.34.56.78>;tag=as4211c518
Call-ID: 3ce1ea71543d92630ae334c43ce5f80f@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '3ce1ea71543d92630ae334c43ce5f80f@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK74408c15;received=87.65.43.21
From: "Unknown" <sip:6666666@87.65.43.21>;tag=as32f65233
To: <sip:12.34.56.78>;tag=as28c5c91e
Call-ID: 295331a045b6fdd71df4d201567ab40c@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '295331a045b6fdd71df4d201567ab40c@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:12.34.56.78:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 87.65.43.21:5060;branch=z9hG4bK409385ca;received=87.65.43.21
From: "Unknown" <sip:7777777@87.65.43.21>;tag=as348c548a
To: <sip:12.34.56.78>;tag=as51b5457d
Call-ID: 4e2808c70ce9272f5e71383a23bae10b@87.65.43.21:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:12.34.56.78>
Accept: application/sdp
Content-Length: 0

<------------->
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: --- (12 headers 0 lines) ---
[Nov 20 10:32:20] VERBOSE[100192] chan_sip.c: Really destroying SIP dialog '4e2808c70ce9272f5e71383a23bae10b@87.65.43.21:5060' Method: OPTIONS
[Nov 20 10:32:22] VERBOSE[100192] chan_sip.c: 
<--- SIP read from UDP:10.1.1.15:5060 --->
INVITE sip:711@10.1.1.220;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 10.1.1.15:5060;branch=z9hG4bK2dff95954cab292c
From: "612" <sip:612@10.1.1.220>;tag=3123b1fc-684847
To: <sip:711@10.1.1.220;user=phone>
Call-ID: 1B12-1220-46684847C5798D30C0F8-003@SipHost
CSeq:3 INVITE
Contact: <sip:612@10.1.1.15:5060>
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent:dlink 12-3856-2886-0.10.50.1-DSLX
Content-Type: application/sdp
Content-Length: 307

v=0
o=612 1792042760 1792042760 IN IP4 10.1.1.15
s=Session SDP
c=IN IP4 10.1.1.15
t=0 0
m=audio 9002 RTP/AVP 8 4 18 2 0
a=rtpmap:8 PCMA/8000
a=fmtp:8 vad=no
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=fmtp:0 vad=no
a=sendrecv
<------------->
