[Sep 28 16:04:47] Connected to Asterisk 1.6.2.0-rc1 currently running on asterisk (pid = 10927) Verbosity is at least 50 Core debug is at least 50 asterisk*CLI> sip set history on SIP History Recording Enabled (use 'sip show history') [2009-09-28 16:04:54] NOTICE[10954]: chan_sip.c:10928 sip_reregister: -- Re-registration for ocY97PJM89@jfk-primary.voicepulse.com [2009-09-28 16:04:54] > doing dnsmgr_lookup for 'jfk-primary.voicepulse.com' [2009-09-28 16:04:54] REGISTER 12 headers, 0 lines [2009-09-28 16:04:54] Reliably Transmitting (no NAT) to 64.61.93.190:5060: REGISTER sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK16181850;rport Max-Forwards: 70 From: ;tag=as7b744049 To: Call-ID: 198c3f7d62d2e5373ecabc2d33b14856@64.105.202.244 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.6.2.0-rc1 Authorization: Digest username="ocY97PJM89", realm="jfk-primary.voicepulse.com", algorithm=MD5, uri="sip:jfk-primary.voicepulse.com", nonce="4ac117a8ce53092fd5e01cc924510a30969a4508", response="dabdb8aac74fb467b5946c4218caa856", qop=auth, cnonce="1205c542", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 16:04:54] NOTICE[10954]: chan_sip.c:10928 sip_reregister: -- Re-registration for ocY97PJM89@jfk-backup.voicepulse.com [2009-09-28 16:04:54] > doing dnsmgr_lookup for 'jfk-backup.voicepulse.com' [2009-09-28 16:04:54] REGISTER 12 headers, 0 lines [2009-09-28 16:04:54] Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK0658c5d5;rport Max-Forwards: 70 From: ;tag=as1951e51a To: Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.6.2.0-rc1 Authorization: Digest username="ocY97PJM89", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ac116900000236ea4f43e6d39dece8913556729e1f47bf6", response="d6453b0b756c2ce3e6542a3f654c968b", qop=auth, cnonce="4964cc23", nc=00000002 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 16:04:54] d <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK16181850;rport=5060 From: ;tag=as7b744049 To: Call-ID: 198c3f7d62d2e5373ecabc2d33b14856@64.105.202.244 CSeq: 104 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (8 headers 0 lines) --- [2009-09-28 16:04:54] d <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK16181850;rport=5060 From: ;tag=as7b744049 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.e6c8 Call-ID: 198c3f7d62d2e5373ecabc2d33b14856@64.105.202.244 CSeq: 104 REGISTER Contact: ;expires=120;received="sip:64.105.202.244:5060" Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (9 headers 0 lines) --- [2009-09-28 16:04:54] Scheduling destruction of SIP dialog '198c3f7d62d2e5373ecabc2d33b14856@64.105.202.244' in 32000 ms (Method: REGISTER) [2009-09-28 16:04:54] NOTICE[10954]: chan_sip.c:17601 handle_response_register: Outbound Registration: Expiry for jfk-primary.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) [2009-09-28 16:04:54] d <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK0658c5d5;rport=5060 From: ;tag=as1951e51a To: Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 104 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (8 headers 0 lines) --- [2009-09-28 16:04:54] d <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK0658c5d5;rport=5060 From: ;tag=as1951e51a To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ebe8 Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 104 REGISTER WWW-Authenticate: Digest realm="jfk-backup.voicepulse.com", nonce="4ac116fa0000309605e81887654387106f411a094588ea1f", qop="auth", stale=true Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (9 headers 0 lines) --- [2009-09-28 16:04:54] Responding to challenge, registration to domain/host name jfk-backup.voicepulse.com [2009-09-28 16:04:54] > doing dnsmgr_lookup for 'jfk-backup.voicepulse.com' [2009-09-28 16:04:54] REGISTER 12 headers, 0 lines [2009-09-28 16:04:54] Reliably Transmitting (no NAT) to 67.108.9.165:5060: REGISTER sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK563090fd;rport Max-Forwards: 70 From: ;tag=as064a2066 To: Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 105 REGISTER User-Agent: Asterisk PBX 1.6.2.0-rc1 Authorization: Digest username="ocY97PJM89", realm="jfk-backup.voicepulse.com", algorithm=MD5, uri="sip:jfk-backup.voicepulse.com", nonce="4ac116fa0000309605e81887654387106f411a094588ea1f", response="6ad603addee6ff0057a0f78ba5808c83", qop=auth, cnonce="08b45e91", nc=00000001 Expires: 120 Contact: Content-Length: 0 --- [2009-09-28 16:04:54] d <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 100 Trying Registration Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK563090fd;rport=5060 From: ;tag=as064a2066 To: Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 105 REGISTER Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (8 headers 0 lines) --- [2009-09-28 16:04:54] de <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK563090fd;rport=5060 From: ;tag=as064a2066 To: ;tag=c97b4d1cb1f3d0da549e06a8d482ef63.0d0f Call-ID: 48de9c726d1482b85ee09a634e866449@64.105.202.244 CSeq: 105 REGISTER Contact: ;expires=120;received="sip:64.105.202.244:5060", ;expires=8;received="sip:64.105.202.244:5060" Server: OpenSIPS (1.4.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:04:54] --- (9 headers 0 lines) --- [2009-09-28 16:04:54] Scheduling destruction of SIP dialog '48de9c726d1482b85ee09a634e866449@64.105.202.244' in 32000 ms (Method: REGISTER) [2009-09-28 16:04:54] NOTICE[10954]: chan_sip.c:17601 handle_response_register: Outbound Registration: Expiry for jfk-backup.voicepulse.com is 120 sec (Scheduling reregistration in 105 s) asterisk*CLI> sip set debug on SIP Debugging re-enabled asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> asterisk*CLI> [2009-09-28 16:05:05] <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-5b5e9146 From: CrossComm, Inc. ;tag=4fdd06ea9dc60c0d To: CrossComm, Inc. ;tag=as35894806 Call-ID: d8d881ee-31b72cd9@192.168.0.30 CSeq: 21331 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="50424a67",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="609aeab1fda1388c708673839efbe578" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:05:05] --- (14 headers 0 lines) --- [2009-09-28 16:05:05] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-28 16:05:05] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-5b5e9146;received=192.168.0.30 From: CrossComm, Inc. ;tag=4fdd06ea9dc60c0d To: CrossComm, Inc. ;tag=as35894806 Call-ID: d8d881ee-31b72cd9@192.168.0.30 CSeq: 21331 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:05] Really destroying SIP dialog 'd8d881ee-31b72cd9@192.168.0.30' Method: SUBSCRIBE [2009-09-28 16:05:08] Reliably Transmitting (no NAT) to 192.168.0.30:5060: OPTIONS sip:line3@192.168.0.30:5060 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK0d10e5dd;rport Max-Forwards: 70 From: "asterisk" ;tag=as7a8f5cbb To: Contact: Call-ID: 17911d01340aee783a262fb5796798be@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Mon, 28 Sep 2009 20:05:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:05:08] <--- SIP read from UDP:192.168.0.30:5060 ---> SIP/2.0 486 Busy Here To: ;tag=f76be8be46253182i0 From: "asterisk" ;tag=as7a8f5cbb Call-ID: 17911d01340aee783a262fb5796798be@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK0d10e5dd Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-28 16:05:08] --- (10 headers 0 lines) --- [2009-09-28 16:05:08] Really destroying SIP dialog '17911d01340aee783a262fb5796798be@64.105.202.244' Method: OPTIONS [2009-09-28 16:05:08] Reliably Transmitting (no NAT) to 64.61.93.190:5060: OPTIONS sip:jfk-primary.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK7026749d;rport Max-Forwards: 70 From: "asterisk" ;tag=as1c7e063d To: Contact: Call-ID: 49d6c7ea4ea9f2b61877f58361890828@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Mon, 28 Sep 2009 20:05:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:05:08] Reliably Transmitting (no NAT) to 192.168.0.30:5061: OPTIONS sip:line4@192.168.0.30:5061 SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK71e0946a;rport Max-Forwards: 70 From: "asterisk" ;tag=as4afe7ad8 To: Contact: Call-ID: 2322f940758eadf20bb157034982b23b@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Mon, 28 Sep 2009 20:05:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:05:08] Reliably Transmitting (no NAT) to 67.108.9.165:5060: OPTIONS sip:jfk-backup.voicepulse.com SIP/2.0 Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK72f9ed43;rport Max-Forwards: 70 From: "asterisk" ;tag=as55e43074 To: Contact: Call-ID: 0b8f17b02c6a1f9b3e3d614269cabd09@64.105.202.244 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.0-rc1 Date: Mon, 28 Sep 2009 20:05:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [2009-09-28 16:05:08] <--- SIP read from UDP:192.168.0.30:5061 ---> SIP/2.0 200 OK To: ;tag=3de4cede920481e2i1 From: "asterisk" ;tag=as4afe7ad8 Call-ID: 2322f940758eadf20bb157034982b23b@64.105.202.244 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK71e0946a Server: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces <-------------> [2009-09-28 16:05:08] --- (10 headers 0 lines) --- [2009-09-28 16:05:08] Really destroying SIP dialog '2322f940758eadf20bb157034982b23b@64.105.202.244' Method: OPTIONS [2009-09-28 16:05:08] <--- SIP read from UDP:64.61.93.190:5060 ---> SIP/2.0 200 OK to keepalive Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK7026749d;rport=5060 From: "asterisk" ;tag=as1c7e063d To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.d2cc Call-ID: 49d6c7ea4ea9f2b61877f58361890828@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:05:08] --- (8 headers 0 lines) --- [2009-09-28 16:05:08] Really destroying SIP dialog '49d6c7ea4ea9f2b61877f58361890828@64.105.202.244' Method: OPTIONS [2009-09-28 16:05:09] <--- SIP read from UDP:67.108.9.165:5060 ---> SIP/2.0 200 OK to keepalive Via: SIP/2.0/UDP 64.105.202.244:5060;branch=z9hG4bK72f9ed43;rport=5060 From: "asterisk" ;tag=as55e43074 To: ;tag=329cfeaa6ded039da25ff8cbb8668bd2.6683 Call-ID: 0b8f17b02c6a1f9b3e3d614269cabd09@64.105.202.244 CSeq: 102 OPTIONS Server: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:05:09] --- (8 headers 0 lines) --- [2009-09-28 16:05:09] Really destroying SIP dialog '0b8f17b02c6a1f9b3e3d614269cabd09@64.105.202.244' Method: OPTIONS [2009-09-28 16:05:13] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-170b7a56 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 101 INVITE Max-Forwards: 70 Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 1132254 1132254 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-28 16:05:13] --- (14 headers 18 lines) --- [2009-09-28 16:05:13] == Using SIP RTP TOS bits 16 [2009-09-28 16:05:13] == Using SIP RTP CoS mark 5 [2009-09-28 16:05:13] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:05:13] Using INVITE request as basis request - 71cca8cf-7fb1ceda@192.168.0.30 [2009-09-28 16:05:13] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 16:05:13] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-170b7a56;received=192.168.0.30 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as0417adaf Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 101 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk.crosscomm.net", nonce="2cff811c" Content-Length: 0 <------------> [2009-09-28 16:05:13] Scheduling destruction of SIP dialog '71cca8cf-7fb1ceda@192.168.0.30' in 6400 ms (Method: INVITE) [2009-09-28 16:05:13] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-170b7a56 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as0417adaf Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 101 ACK Max-Forwards: 70 Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:05:13] --- (10 headers 0 lines) --- [2009-09-28 16:05:13] <--- SIP read from UDP:192.168.0.30:5060 ---> INVITE sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2cff811c",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="1df0c94946bd1e4a53bd9dfc5fe40d30" Contact: CrossComm, Inc. Expires: 240 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 397 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 1132254 1132254 IN IP4 192.168.0.30 s=- c=IN IP4 192.168.0.30 t=0 0 m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2009-09-28 16:05:13] --- (15 headers 18 lines) --- [2009-09-28 16:05:13] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:05:13] Using INVITE request as basis request - 71cca8cf-7fb1ceda@192.168.0.30 [2009-09-28 16:05:13] Found peer 'line3' for 'line3' from 192.168.0.30:5060 [2009-09-28 16:05:13] Found RTP audio format 0 [2009-09-28 16:05:13] Found RTP audio format 2 [2009-09-28 16:05:13] Found RTP audio format 4 [2009-09-28 16:05:13] Found RTP audio format 8 [2009-09-28 16:05:13] Found RTP audio format 18 [2009-09-28 16:05:13] Found RTP audio format 96 [2009-09-28 16:05:13] Found RTP audio format 97 [2009-09-28 16:05:13] Found RTP audio format 98 [2009-09-28 16:05:13] Found RTP audio format 101 [2009-09-28 16:05:13] Peer audio RTP is at port 192.168.0.30:16458 [2009-09-28 16:05:13] Found audio description format PCMU for ID 0 [2009-09-28 16:05:13] Found audio description format G726-32 for ID 2 [2009-09-28 16:05:13] Found audio description format G723 for ID 4 [2009-09-28 16:05:13] Found audio description format PCMA for ID 8 [2009-09-28 16:05:13] Found audio description format G729a for ID 18 [2009-09-28 16:05:13] Found audio description format G726-40 for ID 96 [2009-09-28 16:05:13] Found audio description format G726-24 for ID 97 [2009-09-28 16:05:13] Found audio description format G726-16 for ID 98 [2009-09-28 16:05:13] Found audio description format telephone-event for ID 101 [2009-09-28 16:05:13] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) [2009-09-28 16:05:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-28 16:05:13] Peer audio RTP is at port 192.168.0.30:16458 [2009-09-28 16:05:13] Looking for 8000 in softphones (domain asterisk.crosscomm.net) [2009-09-28 16:05:13] list_route: hop: [2009-09-28 16:05:13] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246;received=192.168.0.30 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-28 16:05:13] -- Executing [8000@softphones:1] ConfBridge("SIP/line3-10346de8", "8000") in new stack [2009-09-28 16:05:13] Audio is at 64.105.202.244 port 20004 [2009-09-28 16:05:13] Adding codec 0x4 (ulaw) to SDP [2009-09-28 16:05:13] Adding codec 0x8 (alaw) to SDP [2009-09-28 16:05:13] Adding non-codec 0x1 (telephone-event) to SDP [2009-09-28 16:05:13] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246;received=192.168.0.30 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as572ed711 Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 295 v=0 o=root 1035845863 1035845863 IN IP4 64.105.202.244 s=Asterisk PBX 1.6.2.0-rc1 c=IN IP4 64.105.202.244 t=0 0 m=audio 20004 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-28 16:05:13] -- Playing 'conf-onlyperson.slin' (language 'en') [2009-09-28 16:05:20] <--- SIP read from UDP:64.61.93.190:5060 ---> INVITE sip:19192460171@64.105.202.244:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKd519.dec698b2.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK582d79fb;rport=5060 From: "Martin Brendan " ;tag=as43129b19 To: Contact: Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 68 Remote-Party-ID: "Martin Brendan " ;privacy=off;screen=no Date: Mon, 28 Sep 2009 20:05:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 410 v=0 o=root 29667 29667 IN IP4 64.61.93.170 s=session c=IN IP4 64.61.93.170 t=0 0 m=audio 19878 RTP/AVP 0 8 3 97 111 5 7 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:7 LPC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [2009-09-28 16:05:20] --- (18 headers 19 lines) --- [2009-09-28 16:05:20] == Using SIP RTP TOS bits 16 [2009-09-28 16:05:20] == Using SIP RTP CoS mark 5 [2009-09-28 16:05:20] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 16:05:20] Using INVITE request as basis request - 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 [2009-09-28 16:05:20] Found peer 'VoicePulse-Primary-Brendan' for '8479227343' from 64.61.93.190:5060 [2009-09-28 16:05:20] Found RTP audio format 0 [2009-09-28 16:05:20] Found RTP audio format 8 [2009-09-28 16:05:20] Found RTP audio format 3 [2009-09-28 16:05:20] Found RTP audio format 97 [2009-09-28 16:05:20] Found RTP audio format 111 [2009-09-28 16:05:20] Found RTP audio format 5 [2009-09-28 16:05:20] Found RTP audio format 7 [2009-09-28 16:05:20] Found RTP audio format 101 [2009-09-28 16:05:20] Peer audio RTP is at port 64.61.93.170:19878 [2009-09-28 16:05:20] Found audio description format PCMU for ID 0 [2009-09-28 16:05:20] Found audio description format PCMA for ID 8 [2009-09-28 16:05:20] Found audio description format GSM for ID 3 [2009-09-28 16:05:20] Found audio description format iLBC for ID 97 [2009-09-28 16:05:20] Found audio description format G726-32 for ID 111 [2009-09-28 16:05:20] Found audio description format DVI4 for ID 5 [2009-09-28 16:05:20] Found audio description format LPC for ID 7 [2009-09-28 16:05:20] Found audio description format telephone-event for ID 101 [2009-09-28 16:05:20] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xcae (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [2009-09-28 16:05:20] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-28 16:05:20] Peer audio RTP is at port 64.61.93.170:19878 [2009-09-28 16:05:20] Looking for 19192460171 in inbound (domain 64.105.202.244) [2009-09-28 16:05:20] list_route: hop: [2009-09-28 16:05:20] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKd519.dec698b2.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK582d79fb;rport=5060 Record-Route: From: "Martin Brendan " ;tag=as43129b19 To: Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [2009-09-28 16:05:20] -- Executing [19192460171@inbound:1] ConfBridge("SIP/VoicePulse-Primary-Brendan-10345ec8", "8000,mMsc") in new stack [2009-09-28 16:05:20] Audio is at 64.105.202.244 port 20012 [2009-09-28 16:05:20] Adding codec 0x2 (gsm) to SDP [2009-09-28 16:05:20] Adding codec 0x4 (ulaw) to SDP [2009-09-28 16:05:20] Adding codec 0x8 (alaw) to SDP [2009-09-28 16:05:20] Adding non-codec 0x1 (telephone-event) to SDP [2009-09-28 16:05:20] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKd519.dec698b2.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK582d79fb;rport=5060 Record-Route: From: "Martin Brendan " ;tag=as43129b19 To: ;tag=as057e6768 Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 316 v=0 o=root 240937752 240937752 IN IP4 64.105.202.244 s=Asterisk PBX 1.6.2.0-rc1 c=IN IP4 64.105.202.244 t=0 0 m=audio 20012 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-28 16:05:20] -- Playing 'conf-onlyone.slin' (language 'en') asterisk*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry 64.61.93.190 ocY97PJM89 198c3f7d62d2e53 0x0 (nothing) No 67.108.9.165 ocY97PJM89 48de9c726d1482b 0x0 (nothing) No 64.61.93.190 ocY97PJM89 68a6553833bee4a 0x4 (ulaw) No Rx: INVITE 192.168.0.30 line3 71cca8cf-7fb1ce 0x4 (ulaw) No Rx: INVITE 4 active SIP dialogs [2009-09-28 16:05:26] Really destroying SIP dialog '198c3f7d62d2e5373ecabc2d33b14856@64.105.202.244' Method: REGISTER [2009-09-28 16:05:26] Really destroying SIP dialog '48de9c726d1482b85ee09a634e866449@64.105.202.244' Method: REGISTER asterisk*CLI> sip show history 71cca8cf-7fb1ce asterisk*CLI> * SIP Call 1. Rx INVITE / 101 INVITE / sip:8000@asterisk.crosscomm.net 2. AuthChal Auth challenge sent for - nc 0 3. TxRespRel SIP/2.0 / 101 INVITE - 401 Unauthorized 4. SchedDestroy 6400 ms 5. Rx ACK / 101 ACK / sip:8000@asterisk.crosscomm.net 6. Rx INVITE / 102 INVITE / sip:8000@asterisk.crosscomm.net 7. CancelDestroy 8. Invite New call: 71cca8cf-7fb1ceda@192.168.0.30 9. AuthOK Auth challenge succesful for line3 10. NewChan Channel SIP/line3-10346de8 - from 71cca8cf-7fb1ceda@192.168.0.3 11. TxResp SIP/2.0 / 102 INVITE - 100 Trying 12. TxResp SIP/2.0 / 102 INVITE - 183 Session Progress [2009-09-28 16:05:35] <--- SIP read from UDP:192.168.0.30:5060 ---> SUBSCRIBE sip:asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-bc599e2a From: CrossComm, Inc. ;tag=8e8065fa1f9bc3f6 To: CrossComm, Inc. ;tag=as35894806 Call-ID: e2f2414a-de7b20e6@192.168.0.30 CSeq: 17435 SUBSCRIBE Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="50424a67",uri="sip:asterisk.crosscomm.net",algorithm=MD5,response="609aeab1fda1388c708673839efbe578" Contact: CrossComm, Inc. Accept: application/simple-message-summary Expires: 2147483647 Event: message-summary User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:05:35] --- (14 headers 0 lines) --- [2009-09-28 16:05:35] Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again. [2009-09-28 16:05:35] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 481 Subscription does not exist Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-bc599e2a;received=192.168.0.30 From: CrossComm, Inc. ;tag=8e8065fa1f9bc3f6 To: CrossComm, Inc. ;tag=as35894806 Call-ID: e2f2414a-de7b20e6@192.168.0.30 CSeq: 17435 SUBSCRIBE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:35] Really destroying SIP dialog 'e2f2414a-de7b20e6@192.168.0.30' Method: SUBSCRIBE asterisk*CLI> sip show history 68a6553833bee4a asterisk*CLI> * SIP Call 1. Rx INVITE / 102 INVITE / sip:19192460171@64.105.202.244:5060 2. NewChan Channel SIP/VoicePulse-Primary-Brendan-10345ec8 - from 68a65538 3. TxResp SIP/2.0 / 102 INVITE - 100 Trying 4. TxResp SIP/2.0 / 102 INVITE - 183 Session Progress asterisk*CLI> sip show history 48de9c726d1482b No such SIP Call ID starting with '48de9c726d1482b' asterisk*CLI> asterisk*CLI> [2009-09-28 16:05:56] <--- SIP read from UDP:64.61.93.190:5060 ---> CANCEL sip:19192460171@64.105.202.244:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0 From: "Martin Brendan " ;tag=as43129b19 Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 To: CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:05:56] --- (9 headers 0 lines) --- [2009-09-28 16:05:56] Sending to 64.61.93.190 : 5060 (no NAT) [2009-09-28 16:05:56] <--- Reliably Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0;received=64.61.93.190 Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKd519.dec698b2.0 Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK582d79fb;rport=5060 From: "Martin Brendan " ;tag=as43129b19 To: ;tag=as057e6768 Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:56] <--- Transmitting (no NAT) to 64.61.93.190:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0;received=64.61.93.190 From: "Martin Brendan " ;tag=as43129b19 To: ;tag=as057e6768 Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 CSeq: 102 CANCEL Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:56] <--- SIP read from UDP:64.61.93.190:5060 ---> ACK sip:19192460171@64.105.202.244:5060 SIP/2.0 Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKd519.d0bdc621.0 From: "Martin Brendan " ;tag=as43129b19 Call-ID: 68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170 To: ;tag=as057e6768 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (i386/linux)) Content-Length: 0 <-------------> [2009-09-28 16:05:56] --- (9 headers 0 lines) --- [2009-09-28 16:05:56] Really destroying SIP dialog '68a6553833bee4a0093eb6ca721ab6ba@64.61.93.170' Method: ACK [2009-09-28 16:05:56] <--- SIP read from UDP:192.168.0.30:5060 ---> CANCEL sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 CANCEL Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2cff811c",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="04fef36f201cdcc96d083b611ae379af" User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:05:56] --- (10 headers 0 lines) --- [2009-09-28 16:05:56] Sending to 192.168.0.30 : 5060 (no NAT) [2009-09-28 16:05:56] <--- Reliably Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246;received=192.168.0.30 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as572ed711 Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 INVITE Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:56] <--- Transmitting (no NAT) to 192.168.0.30:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246;received=192.168.0.30 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as572ed711 Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 CANCEL Server: Asterisk PBX 1.6.2.0-rc1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-28 16:05:56] <--- SIP read from UDP:192.168.0.30:5060 ---> ACK sip:8000@asterisk.crosscomm.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.30:5060;branch=z9hG4bK-2137246 From: CrossComm, Inc. ;tag=e3fe684b98b0b58eo0 To: ;tag=as572ed711 Call-ID: 71cca8cf-7fb1ceda@192.168.0.30 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="line3",realm="asterisk.crosscomm.net",nonce="2cff811c",uri="sip:8000@asterisk.crosscomm.net",algorithm=MD5,response="1df0c94946bd1e4a53bd9dfc5fe40d30" Contact: CrossComm, Inc. User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 <-------------> [2009-09-28 16:05:56] --- (11 headers 0 lines) --- [2009-09-28 16:05:57] Really destroying SIP dialog '71cca8cf-7fb1ceda@192.168.0.30' Method: ACK