hostname*CLI> <--- SIP read from UDP:aa.aa.56.45:2479 ---> INVITE sip:1000103@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK6f1593dc9A668DA5 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: CSeq: 1 INVITE Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1511749211 1511749211 IN IP4 bb.bb.3.246 s=Polycom IP Phone c=IN IP4 bb.bb.3.246 t=0 0 a=sendrecv m=audio 2234 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (15 headers 13 lines) --- Sending to aa.aa.56.45:2479 (NAT) Sending to aa.aa.56.45:2479 (NAT) Using INVITE request as basis request - fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Found peer '1000105' for '1000105' from aa.aa.56.45:2479 <--- Reliably Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK6f1593dc9A668DA5;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as593e13d0 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 1 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50b56a13" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'fe4426d0-2464b6c1-941646e4@bb.bb.3.246' in 32000 ms (Method: INVITE) <--- SIP read from UDP:aa.aa.56.45:2479 ---> ACK sip:1000103@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK6f1593dc9A668DA5 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as593e13d0 CSeq: 1 ACK Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:2479 ---> INVITE sip:1000103@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK661413d95CAD918 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: CSeq: 2 INVITE Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="1000105", realm="asterisk", nonce="50b56a13", uri="sip:1000103@hostname.example.com;user=phone", response="1247718ff32366b200808466c0aa3458", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1511749211 1511749211 IN IP4 bb.bb.3.246 s=Polycom IP Phone c=IN IP4 bb.bb.3.246 t=0 0 a=sendrecv m=audio 2234 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (16 headers 13 lines) --- Sending to aa.aa.56.45:2479 (NAT) Using INVITE request as basis request - fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Found peer '1000105' for '1000105' from aa.aa.56.45:2479 == Using SIP RTP CoS mark 5 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g722|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port bb.bb.3.246:2234 Looking for 1000103 in from-internal (domain hostname.example.com) sip_route_dump: route/path hop: <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK661413d95CAD918;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [1000103@from-internal:1] NoOp("SIP/1000105-0000001c", "Start call 1511749212.44") in new stack -- Executing [1000103@from-internal:2] Dial("SIP/1000105-0000001c", "SIP/1000103,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 15450 Adding codec g722 to SDP Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to aa.aa.56.45:5060: INVITE sip:1000103@aa.aa.56.45:5060;line=66b08f90bcd8652 SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK0e4e36d7;rport Max-Forwards: 70 From: "1000105" ;tag=as0bd2e537 To: Contact: Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 102 INVITE User-Agent: Switch Date: Mon, 27 Nov 2017 02:20:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 321 v=0 o=root 247520448 247520448 IN IP4 cc.cc.66.251 s=Asterisk PBX 13.17.2 c=IN IP4 cc.cc.66.251 t=0 0 m=audio 15450 RTP/AVP 9 0 8 3 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/1000103 <--- SIP read from UDP:aa.aa.56.45:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK0e4e36d7;rport=5060 From: "1000105" ;tag=as0bd2e537 To: Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 102 INVITE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:5060 ---> jaK <-------------> <--- SIP read from UDP:aa.aa.56.45:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK0e4e36d7;rport=5060 From: "1000105" ;tag=as0bd2e537 To: ;tag=617246271 Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/1000103-0000001d is ringing <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK661413d95CAD918;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:dd.dd.250.16:5060 ---> jaK <-------------> <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK6e138e55ED9FCFE0 From: "1000105" ;tag=8FBD8C91-C807A954 To: CSeq: 1 SUBSCRIBE Call-ID: ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to aa.aa.56.45:2479 (NAT) Creating new subscription Sending to aa.aa.56.45:2479 (NAT) sip_route_dump: route/path hop: Found peer '1000105' for '1000105' from aa.aa.56.45:2479 <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK6e138e55ED9FCFE0;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=8FBD8C91-C807A954 To: ;tag=as149c9f90 Call-ID: ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246 CSeq: 1 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="79f7ddd0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bKfd595328C1004F49 From: "1000105" ;tag=8FBD8C91-C807A954 To: CSeq: 2 SUBSCRIBE Call-ID: ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="1000105", realm="asterisk", nonce="79f7ddd0", uri="sip:1000105@hostname.example.com", response="dee068f04c8eab59226dd8a2ab44d412", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Creating new subscription Sending to aa.aa.56.45:2479 (NAT) Found peer '1000105' for '1000105' from aa.aa.56.45:2479 Looking for 1000105 in from-internal (domain hostname.example.com) <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bKfd595328C1004F49;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=8FBD8C91-C807A954 To: ;tag=as149c9f90 Call-ID: ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246 CSeq: 2 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'ea7cd6f1-aeb2a674-f73d0ced@bb.bb.3.246' Method: SUBSCRIBE <--- SIP read from UDP:aa.aa.56.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK0e4e36d7;rport=5060 From: "1000105" ;tag=as0bd2e537 To: ;tag=617246271 Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 185 v=0 o=1000103 642 4017 IN IP4 bb.bb.1.252 s=Talk c=IN IP4 bb.bb.1.252 t=0 0 m=audio 7078 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (10 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port bb.bb.1.252:7078 sip_route_dump: route/path hop: Transmitting (NAT) to aa.aa.56.45:5060: ACK sip:david@bb.bb.1.252 SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK6571b4d7;rport Max-Forwards: 70 From: "1000105" ;tag=as0bd2e537 To: ;tag=617246271 Contact: Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 102 ACK User-Agent: Switch Content-Length: 0 --- -- SIP/1000103-0000001d answered SIP/1000105-0000001c Audio is at 14912 Adding codec g722 to SDP Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK661413d95CAD918;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 1400266102 1400266102 IN IP4 cc.cc.66.251 s=Asterisk PBX 13.17.2 c=IN IP4 cc.cc.66.251 t=0 0 m=audio 14912 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/1000103-0000001d joined 'simple_bridge' basic-bridge -- Channel SIP/1000105-0000001c joined 'simple_bridge' basic-bridge <--- SIP read from UDP:aa.aa.56.45:2479 ---> ACK sip:1000103@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bKbb670804CA4B7E9D From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 CSeq: 2 ACK Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:2479 ---> INVITE sip:1000103@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK3bb19f3852C04879 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 CSeq: 3 INVITE Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="1000105", realm="asterisk", nonce="50b56a13", uri="sip:1000103@hostname.example.com;user=phone", response="1247718ff32366b200808466c0aa3458", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 211 v=0 o=- 1511749211 1511749212 IN IP4 bb.bb.3.246 s=Polycom IP Phone c=IN IP4 bb.bb.3.246 t=0 0 a=sendonly m=audio 2234 RTP/AVP 9 127 a=sendonly a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 <-------------> --- (16 headers 10 lines) --- Sending to aa.aa.56.45:2479 (NAT) Found RTP audio format 9 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format telephone-event for ID 127 Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port bb.bb.3.246:2234 <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK3bb19f3852C04879;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 3 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 14912 Adding codec g722 to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK3bb19f3852C04879;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 3 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=root 1400266102 1400266103 IN IP4 cc.cc.66.251 s=Asterisk PBX 13.17.2 c=IN IP4 cc.cc.66.251 t=0 0 m=audio 14912 RTP/AVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=recvonly <------------> -- Started music on hold, class 'default', on channel 'SIP/1000103-0000001d' <--- SIP read from UDP:aa.aa.56.45:2479 ---> ACK sip:1000103@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK29c19d8cA14F21B5 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 CSeq: 3 ACK Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:2479 ---> INVITE sip:1000102@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK2fb7564dC86FFD48 From: "1000105" ;tag=316AF951-6B45AB94 To: CSeq: 1 INVITE Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1511749220 1511749220 IN IP4 bb.bb.3.246 s=Polycom IP Phone c=IN IP4 bb.bb.3.246 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (15 headers 13 lines) --- Sending to aa.aa.56.45:2479 (NAT) Sending to aa.aa.56.45:2479 (NAT) Using INVITE request as basis request - 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Found peer '1000105' for '1000105' from aa.aa.56.45:2479 <--- Reliably Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK2fb7564dC86FFD48;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=316AF951-6B45AB94 To: ;tag=as7c2205e0 Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 CSeq: 1 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0fdb982b" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3d5217a9-b30cba1c-fff33465@bb.bb.3.246' in 32000 ms (Method: INVITE) <--- SIP read from UDP:aa.aa.56.45:2479 ---> ACK sip:1000102@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK2fb7564dC86FFD48 From: "1000105" ;tag=316AF951-6B45AB94 To: ;tag=as7c2205e0 CSeq: 1 ACK Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:2479 ---> INVITE sip:1000102@hostname.example.com;user=phone SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK95e3f71070D27B81 From: "1000105" ;tag=316AF951-6B45AB94 To: CSeq: 2 INVITE Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="1000105", realm="asterisk", nonce="0fdb982b", uri="sip:1000102@hostname.example.com;user=phone", response="4daea05397daaa3912a36462c6f45e64", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1511749220 1511749220 IN IP4 bb.bb.3.246 s=Polycom IP Phone c=IN IP4 bb.bb.3.246 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (16 headers 13 lines) --- Sending to aa.aa.56.45:2479 (NAT) Using INVITE request as basis request - 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Found peer '1000105' for '1000105' from aa.aa.56.45:2479 == Using SIP RTP CoS mark 5 Found RTP audio format 9 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 127 Found audio description format G722 for ID 9 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g722|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port bb.bb.3.246:2236 Looking for 1000102 in from-internal (domain hostname.example.com) sip_route_dump: route/path hop: <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK95e3f71070D27B81;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=316AF951-6B45AB94 To: Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [1000102@from-internal:1] NoOp("SIP/1000105-0000001e", "Start call 1511749221.46") in new stack -- Executing [1000102@from-internal:2] Dial("SIP/1000105-0000001e", "SIP/1000102,60") in new stack == Using SIP RTP CoS mark 5 Audio is at 12896 Adding codec g722 to SDP Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to aa.aa.56.45:45201: INVITE sip:1000102@aa.aa.56.45:45201;ob SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK3a349362;rport Max-Forwards: 70 From: "1000105" ;tag=as6d4d56a5 To: Contact: Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 CSeq: 102 INVITE User-Agent: Switch Date: Mon, 27 Nov 2017 02:20:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 321 v=0 o=root 233983898 233983898 IN IP4 cc.cc.66.251 s=Asterisk PBX 13.17.2 c=IN IP4 cc.cc.66.251 t=0 0 m=audio 12896 RTP/AVP 9 0 8 3 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv --- -- Called SIP/1000102 <--- SIP read from UDP:aa.aa.56.45:45201 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP cc.cc.66.251:5060;rport=5060;received=cc.cc.66.251;branch=z9hG4bK3a349362 Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 From: "1000105" ;tag=as6d4d56a5 To: CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:45201 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP cc.cc.66.251:5060;rport=5060;received=cc.cc.66.251;branch=z9hG4bK3a349362 Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 From: "1000105" ;tag=as6d4d56a5 To: ;tag=PAVBN2iDDRbpXCs2qbASlUiM6bZmFSTd CSeq: 102 INVITE Contact: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip_route_dump: route/path hop: -- SIP/1000102-0000001f is ringing <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK95e3f71070D27B81;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=316AF951-6B45AB94 To: ;tag=as0cbf0907 Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK4cf8ad58AD767CD9 From: "1000105" ;tag=A9122AF0-1ACAAD21 To: CSeq: 1 SUBSCRIBE Call-ID: 87ec78ac-d151cd15-9a4be820@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to aa.aa.56.45:2479 (NAT) Creating new subscription Sending to aa.aa.56.45:2479 (NAT) sip_route_dump: route/path hop: Found peer '1000105' for '1000105' from aa.aa.56.45:2479 <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK4cf8ad58AD767CD9;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=A9122AF0-1ACAAD21 To: ;tag=as33a3f11d Call-ID: 87ec78ac-d151cd15-9a4be820@bb.bb.3.246 CSeq: 1 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ae3b456" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '87ec78ac-d151cd15-9a4be820@bb.bb.3.246' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bKc4cdf3b1B272B8B4 From: "1000105" ;tag=A9122AF0-1ACAAD21 To: CSeq: 2 SUBSCRIBE Call-ID: 87ec78ac-d151cd15-9a4be820@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="1000105", realm="asterisk", nonce="2ae3b456", uri="sip:1000105@hostname.example.com", response="aaef38a1d8df64b0bdc80b869e4708f9", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Creating new subscription Sending to aa.aa.56.45:2479 (NAT) Found peer '1000105' for '1000105' from aa.aa.56.45:2479 Looking for 1000105 in from-internal (domain hostname.example.com) <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bKc4cdf3b1B272B8B4;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=A9122AF0-1ACAAD21 To: ;tag=as33a3f11d Call-ID: 87ec78ac-d151cd15-9a4be820@bb.bb.3.246 CSeq: 2 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog '87ec78ac-d151cd15-9a4be820@bb.bb.3.246' Method: SUBSCRIBE <--- SIP read from UDP:aa.aa.56.45:45201 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP cc.cc.66.251:5060;rport=5060;received=cc.cc.66.251;branch=z9hG4bK3a349362 Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 From: "1000105" ;tag=as6d4d56a5 To: ;tag=PAVBN2iDDRbpXCs2qbASlUiM6bZmFSTd CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 263 v=0 o=- 3720738020 3720738021 IN IP4 bb.bb.1.213 s=pjmedia c=IN IP4 bb.bb.1.213 t=0 0 m=audio 4024 RTP/AVP 8 101 c=IN IP4 bb.bb.1.213 a=rtcp:4025 IN IP4 bb.bb.1.213 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 12 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port bb.bb.1.213:4024 sip_route_dump: route/path hop: Transmitting (NAT) to aa.aa.56.45:45201: ACK sip:1000102@aa.aa.56.45:45201;ob SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK076c3a9d;rport Max-Forwards: 70 From: "1000105" ;tag=as6d4d56a5 To: ;tag=PAVBN2iDDRbpXCs2qbASlUiM6bZmFSTd Contact: Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 CSeq: 102 ACK User-Agent: Switch Content-Length: 0 --- -- SIP/1000102-0000001f answered SIP/1000105-0000001e Audio is at 15760 Adding codec g722 to SDP Adding codec ulaw to SDP Adding codec alaw to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK95e3f71070D27B81;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=316AF951-6B45AB94 To: ;tag=as0cbf0907 Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 CSeq: 2 INVITE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 298 v=0 o=root 537541103 537541103 IN IP4 cc.cc.66.251 s=Asterisk PBX 13.17.2 c=IN IP4 cc.cc.66.251 t=0 0 m=audio 15760 RTP/AVP 9 0 8 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> -- Channel SIP/1000102-0000001f joined 'simple_bridge' basic-bridge -- Channel SIP/1000105-0000001e joined 'simple_bridge' basic-bridge <--- SIP read from UDP:aa.aa.56.45:2479 ---> ACK sip:1000102@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK3d71abc592DB8B30 From: "1000105" ;tag=316AF951-6B45AB94 To: ;tag=as0cbf0907 CSeq: 2 ACK Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:dd.dd.250.16:5060 ---> jaK <-------------> <--- SIP read from UDP:aa.aa.56.45:2479 ---> REFER sip:1000103@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK58221e1D08DA244 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 CSeq: 4 REFER Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Refer-To: Referred-By: Authorization: Digest username="1000105", realm="asterisk", nonce="50b56a13", uri="sip:1000103@hostname.example.com;user=phone", response="f85f4aa09115da254a25a579c10b1454", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Call fe4426d0-2464b6c1-941646e4@bb.bb.3.246 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 1000102@from-internal by 1000105@hostname.example.com <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK58221e1D08DA244;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 4 REFER Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Channel SIP/1000103-0000001d left 'simple_bridge' basic-bridge -- Channel SIP/1000105-0000001e left 'simple_bridge' basic-bridge -- Channel SIP/1000103-0000001d swapped with SIP/1000105-0000001e into 'simple_bridge' basic-bridge Reliably Transmitting (NAT) to aa.aa.56.45:2479: NOTIFY sip:1000105@bb.bb.3.246 SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK1ac28a19;rport Max-Forwards: 70 From: ;tag=as5e6a1521 To: "1000105" ;tag=9E31F108-FD2ADEE9 Contact: Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 102 NOTIFY User-Agent: Switch Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- -- Channel SIP/1000105-0000001c left 'simple_bridge' basic-bridge == Spawn extension (from-internal, 1000103, 2) exited non-zero on 'SIP/1000105-0000001c' -- Executing [h@from-internal:1] NoOp("SIP/1000105-0000001c", "Hangup call 1511749212.44") in new stack Scheduling destruction of SIP dialog 'fe4426d0-2464b6c1-941646e4@bb.bb.3.246' in 32000 ms (Method: REFER) == Spawn extension (from-internal, 1000102, 2) exited non-zero on 'SIP/1000105-0000001e' -- Executing [h@from-internal:1] NoOp("SIP/1000105-0000001e", "Hangup call 1511749221.46") in new stack Scheduling destruction of SIP dialog '3d5217a9-b30cba1c-fff33465@bb.bb.3.246' in 32000 ms (Method: ACK) Reliably Transmitting (NAT) to aa.aa.56.45:2479: BYE sip:1000105@bb.bb.3.246 SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK5b46c64b;rport Max-Forwards: 70 From: ;tag=as0cbf0907 To: "1000105" ;tag=316AF951-6B45AB94 Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 CSeq: 102 BYE User-Agent: Switch Proxy-Authorization: Digest username="1000105", realm="asterisk", algorithm=MD5, uri="sip:hostname.example.com", nonce="0fdb982b", response="adb938de3183a455d9d182dd23a1c676" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- -- Stopped music on hold on SIP/1000103-0000001d <--- SIP read from UDP:aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK1ac28a19;rport From: ;tag=as5e6a1521 To: "1000105" ;tag=9E31F108-FD2ADEE9 CSeq: 102 NOTIFY Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: Event: refer;id=4 User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:aa.aa.56.45:2479 ---> BYE sip:1000103@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK4d3ec15d81E20378 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 CSeq: 5 BYE Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 Contact: User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Authorization: Digest username="1000105", realm="asterisk", nonce="50b56a13", uri="sip:1000103@hostname.example.com;user=phone", response="458aad70a55da58df19f1b6e05fb7b08", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to aa.aa.56.45:2479 (NAT) Scheduling destruction of SIP dialog 'fe4426d0-2464b6c1-941646e4@bb.bb.3.246' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK4d3ec15d81E20378;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=9E31F108-FD2ADEE9 To: ;tag=as5e6a1521 Call-ID: fe4426d0-2464b6c1-941646e4@bb.bb.3.246 CSeq: 5 BYE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:aa.aa.56.45:2479 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK5b46c64b;rport From: ;tag=as0cbf0907 To: "1000105" ;tag=316AF951-6B45AB94 CSeq: 102 BYE Call-ID: 3d5217a9-b30cba1c-fff33465@bb.bb.3.246 Contact: User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3d5217a9-b30cba1c-fff33465@bb.bb.3.246' Method: ACK <--- SIP read from UDP:aa.aa.56.45:5060 ---> jaK <-------------> hostname*CLI> hostname*CLI> hostname*CLI> hostname*CLI> hostname*CLI> hostname*CLI> Really destroying SIP dialog 'bxFxbxP3uJ@127.0.0.1' Method: OPTIONS Really destroying SIP dialog 'QTCCvEN0j1@127.0.0.1' Method: OPTIONS Really destroying SIP dialog 'CFEXJWfMcC@127.0.0.1' Method: OPTIONS Really destroying SIP dialog 'xuyOrGk6Ca@127.0.0.1' Method: OPTIONS hostname*CLI> core show channels co concise count hostname*CLI> core show channels concise SIP/1000103-0000001d!from-internal!!1!Up!AppDial!(Outgoing Line)!1000103!!!3!22!e1a60992-beec-4fd7-ac56-6694723c1c7d!1511749212.45 SIP/1000102-0000001f!from-internal!!1!Up!AppDial!(Outgoing Line)!1000102!!!3!14!e1a60992-beec-4fd7-ac56-6694723c1c7d!1511749221.47 <--- SIP read from UDP:dd.dd.250.16:5060 ---> jaK <-------------> <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK5eb209391E8ADBCC From: "1000105" ;tag=6EC63809-CAA7193C To: CSeq: 1 SUBSCRIBE Call-ID: e9f19075-8fcb2040-9a6bc611@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (15 headers 0 lines) --- Sending to aa.aa.56.45:2479 (NAT) Creating new subscription Sending to aa.aa.56.45:2479 (NAT) sip_route_dump: route/path hop: Found peer '1000105' for '1000105' from aa.aa.56.45:2479 <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK5eb209391E8ADBCC;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=6EC63809-CAA7193C To: ;tag=as50ca2f36 Call-ID: e9f19075-8fcb2040-9a6bc611@bb.bb.3.246 CSeq: 1 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="62601827" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e9f19075-8fcb2040-9a6bc611@bb.bb.3.246' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:aa.aa.56.45:2479 ---> SUBSCRIBE sip:1000105@hostname.example.com SIP/2.0 Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK316bcdd4ACABB10D From: "1000105" ;tag=6EC63809-CAA7193C To: CSeq: 2 SUBSCRIBE Call-ID: e9f19075-8fcb2040-9a6bc611@bb.bb.3.246 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: presence User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.9.0509 Accept-Language: en Accept: application/xpidf+xml,text/xml+msrtc.pidf Authorization: Digest username="1000105", realm="asterisk", nonce="62601827", uri="sip:1000105@hostname.example.com", response="8b6a9fca22229c8db7a7b5441242c238", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> --- (16 headers 0 lines) --- Creating new subscription Sending to aa.aa.56.45:2479 (NAT) Found peer '1000105' for '1000105' from aa.aa.56.45:2479 Looking for 1000105 in from-internal (domain hostname.example.com) <--- Transmitting (NAT) to aa.aa.56.45:2479 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP bb.bb.3.246;branch=z9hG4bK316bcdd4ACABB10D;received=aa.aa.56.45;rport=2479 From: "1000105" ;tag=6EC63809-CAA7193C To: ;tag=as50ca2f36 Call-ID: e9f19075-8fcb2040-9a6bc611@bb.bb.3.246 CSeq: 2 SUBSCRIBE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> Really destroying SIP dialog 'e9f19075-8fcb2040-9a6bc611@bb.bb.3.246' Method: SUBSCRIBE <--- SIP read from UDP:aa.aa.56.45:5060 ---> jaK <-------------> <--- SIP read from UDP:aa.aa.56.45:45201 ---> BYE sip:1000105@cc.cc.66.251:5060 SIP/2.0 Via: SIP/2.0/UDP bb.bb.1.213:45201;rport;branch=z9hG4bKPjAhWGM88vh.ivAelHoqAq4uRKJ.MAVTYr Max-Forwards: 70 From: ;tag=PAVBN2iDDRbpXCs2qbASlUiM6bZmFSTd To: "1000105" ;tag=as6d4d56a5 Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 CSeq: 22384 BYE User-Agent: CSipSimple_zeroflte-24/r2457 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to aa.aa.56.45:45201 (NAT) Scheduling destruction of SIP dialog '36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060' in 54848 ms (Method: BYE) <--- Transmitting (NAT) to aa.aa.56.45:45201 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP bb.bb.1.213:45201;branch=z9hG4bKPjAhWGM88vh.ivAelHoqAq4uRKJ.MAVTYr;received=aa.aa.56.45;rport=45201 From: ;tag=PAVBN2iDDRbpXCs2qbASlUiM6bZmFSTd To: "1000105" ;tag=as6d4d56a5 Call-ID: 36c47d646fd61c274cfe56ce33ad7afc@cc.cc.66.251:5060 CSeq: 22384 BYE Server: Switch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> -- Channel SIP/1000102-0000001f left 'simple_bridge' basic-bridge -- Channel SIP/1000103-0000001d left 'simple_bridge' basic-bridge Scheduling destruction of SIP dialog '1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060' in 6400 ms (Method: INVITE) Reliably Transmitting (NAT) to aa.aa.56.45:5060: BYE sip:david@bb.bb.1.252 SIP/2.0 Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK48bc00e8;rport Max-Forwards: 70 From: "1000105" ;tag=as0bd2e537 To: ;tag=617246271 Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 103 BYE User-Agent: Switch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:aa.aa.56.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP cc.cc.66.251:5060;branch=z9hG4bK48bc00e8;rport=5060 From: "1000105" ;tag=as0bd2e537 To: ;tag=617246271 Call-ID: 1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060 CSeq: 103 BYE User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '1ec5874c695d880b2822cfe8136a5806@cc.cc.66.251:5060' Method: INVITE hostname*CLI> quit