<-- SIP read from 10.10.30.217:2784: INVITE sip:100@10.10.30.168 SIP/2.0 From: sip:6825@sergei.leapstone.com;tag=1c9901 To: sip:100@10.10.30.168 Call-Id: call-1127825074-1@10.10.30.217 Cseq: 1 INVITE Contact: Content-Type: application/sdp Content-Length: 306 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (VxWorks) Date: Tue, 27 Sep 2005 12:44:38 GMT Via: SIP/2.0/UDP 10.10.30.217 v=0 o=Pingtel 5 5 IN IP4 10.10.30.217 s=phone-call c=IN IP4 10.10.30.217 t=0 0 m=audio 8766 RTP/AVP 96 97 0 8 18 98 a=rtpmap:96 eg711u/8000/1 a=rtpmap:97 eg711a/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000/1 --- (14 headers 13 lines)--- Using INVITE request as basis request - call-1127825074-1@10.10.30.217 Sending to 10.10.30.217 : 5060 (non-NAT) Found user '6825' Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 98 Peer audio RTP is at port 10.10.30.217:8766 Found description format eg711u Found description format eg711a Found description format pcmu Found description format pcma Found description format g729 Found description format telephone-event Capabilities: us - 0x40d (g723|ulaw|alaw|ilbc), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x40c (ulaw|alaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 100 in default list_route: hop: Transmitting (no NAT) to 10.10.30.217:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.30.217 From: sip:6825@sergei.leapstone.com;tag=1c9901 To: sip:100@10.10.30.168 Call-ID: call-1127825074-1@10.10.30.217 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing SIPDtmfMode("SIP/6825-1161", "rfc2833") in new stack -- Executing Dial("SIP/6825-1161", "SIP/100@sokhapkin.com|20") in new stack -- parse_srv: SRV mapped to host sokhapkin.dyndns.org, port 5060 We're at 10.10.30.168 port 14990 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x1 (g723) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 14 lines Reliably Transmitting (no NAT) to 24.149.137.157:5060: INVITE sip:100@sokhapkin.com SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK44d07cb2 From: "6825" ;tag=as3e08a186 To: Contact: Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 27 Sep 2005 12:44:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 316 v=0 o=root 16698 16698 IN IP4 10.10.30.168 s=session c=IN IP4 10.10.30.168 t=0 0 m=audio 14990 RTP/AVP 0 8 18 97 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 100@sokhapkin.com sergei*CLI> <-- SIP read from 192.168.2.161:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK44d07cb2 From: "6825" ;tag=as3e08a186 To: Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 192.168.2.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK44d07cb2 From: "6825" ;tag=as3e08a186 To: ;tag=as1a8543bb Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 294 v=0 o=root 9960 9960 IN IP4 24.149.137.157 s=session c=IN IP4 24.149.137.157 t=0 0 m=audio 15260 RTP/AVP 0 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 13 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 24.149.137.157:15260 Found description format PCMU Found description format PCMA Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0x50d (g723|ulaw|alaw|g729|ilbc), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50c (ulaw|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 24.149.137.157, port 5060 Transmitting (no NAT) to 24.149.137.157:5060: ACK sip:100@24.149.137.157 SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK3921cdba From: "6825" ;tag=as3e08a186 To: ;tag=as1a8543bb Contact: Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/sokhapkin.com-39ce answered SIP/6825-1161 We're at 10.10.30.168 port 19818 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x1 (g723) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 10.10.30.217:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.30.217 From: sip:6825@sergei.leapstone.com;tag=1c9901 To: sip:100@10.10.30.168;tag=as6fa1d68f Call-ID: call-1127825074-1@10.10.30.217 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 16698 16698 IN IP4 10.10.30.168 s=session c=IN IP4 10.10.30.168 t=0 0 m=audio 19818 RTP/AVP 0 8 97 4 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/6825-1161 and SIP/sokhapkin.com-39ce set_destination: Parsing for address/port to send to set_destination: set destination to 24.149.137.157, port 5060 We're at 10.10.30.168 port 14990 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting (no NAT) to 24.149.137.157:5060: INVITE sip:100@24.149.137.157 SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK12c2a4b8 From: "6825" ;tag=as3e08a186 To: ;tag=as1a8543bb Contact: Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 291 v=0 o=root 16698 16699 IN IP4 10.10.30.217 s=session c=IN IP4 10.10.30.217 t=0 0 m=audio 8766 RTP/AVP 0 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- sergei*CLI> <-- SIP read from 192.168.2.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK12c2a4b8 From: "6825" ;tag=as3e08a186 To: ;tag=as1a8543bb Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 294 v=0 o=root 9960 9961 IN IP4 24.149.137.157 s=session c=IN IP4 24.149.137.157 t=0 0 m=audio 15260 RTP/AVP 0 8 18 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 13 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 24.149.137.157:15260 Found description format PCMU Found description format PCMA Found description format G729 Found description format iLBC Found description format telephone-event Capabilities: us - 0x50d (g723|ulaw|alaw|g729|ilbc), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50c (ulaw|alaw|g729|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 24.149.137.157, port 5060 Transmitting (no NAT) to 24.149.137.157:5060: ACK sip:100@24.149.137.157 SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK1f2998f8 From: "6825" ;tag=as3e08a186 To: ;tag=as1a8543bb Contact: Call-ID: 5936877b1a2d8d3c7d31f0077ebc0c59@10.10.30.168 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 --- sergei*CLI> <-- SIP read from 10.10.30.217:2784: ACK sip:100@10.10.30.168 SIP/2.0 Contact: sip:6825@10.10.30.217 From: sip:6825@sergei.leapstone.com;tag=1c9901 To: sip:100@10.10.30.168;tag=as6fa1d68f Call-Id: call-1127825074-1@10.10.30.217 Cseq: 1 ACK Accept-Language: en User-Agent: Pingtel/2.1.11 (VxWorks) Date: Tue, 27 Sep 2005 12:44:40 GMT Via: SIP/2.0/UDP 10.10.30.217 Content-Length: 0 --- (11 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.30.217, port 5060 We're at 10.10.30.168 port 19818 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x400 (ilbc) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting (no NAT) to 10.10.30.217:5060: INVITE sip:6825@10.10.30.217 SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK314091f1;rport From: sip:100@10.10.30.168;tag=as6fa1d68f To: sip:6825@sergei.leapstone.com;tag=1c9901 Contact: Call-ID: call-1127825074-1@10.10.30.217 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 16698 16699 IN IP4 24.149.137.157 s=session c=IN IP4 24.149.137.157 t=0 0 m=audio 15260 RTP/AVP 0 8 97 18 98 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-16 a=silenceSupp:off - - - - --- sergei*CLI> <-- SIP read from 10.10.30.217:2784: SIP/2.0 200 OK From: sip:100@10.10.30.168;tag=as6fa1d68f To: sip:6825@sergei.leapstone.com;tag=1c9901 Call-Id: call-1127825074-1@10.10.30.217 Cseq: 102 INVITE Content-Type: application/sdp Content-Length: 246 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK314091f1;rport=5060 Contact: sip:6825@10.10.30.217 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE User-Agent: Pingtel/2.1.11 (VxWorks) Date: Tue, 27 Sep 2005 12:44:40 GMT v=0 o=Pingtel 5 5 IN IP4 10.10.30.217 s=phone-call c=IN IP4 10.10.30.217 t=0 0 m=audio 8766 RTP/AVP 0 8 18 98 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000/1 --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 98 Peer audio RTP is at port 10.10.30.217:8766 Found description format pcmu Found description format pcma Found description format g729 Found description format telephone-event Capabilities: us - 0x40d (g723|ulaw|alaw|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.30.217, port 5060 Transmitting (no NAT) to 10.10.30.217:5060: ACK sip:6825@10.10.30.217 SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK53d52317;rport From: sip:100@10.10.30.168;tag=as6fa1d68f To: sip:6825@sergei.leapstone.com;tag=1c9901 Contact: Call-ID: call-1127825074-1@10.10.30.217 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- 11 headers, 3 lines Reliably Transmitting (no NAT) to 10.10.30.217:5060: NOTIFY sip:6825@10.10.30.217;LINEID=dd54568275cd918595693024407ca61e SIP/2.0 Via: SIP/2.0/UDP 10.10.30.168:5060;branch=z9hG4bK38f7e52f From: "asterisk" ;tag=as26e81186 To: Contact: Call-ID: 03334ce840ba042e081884300d65aa97@10.10.30.168 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Message-Account: sip:asterisk@ Messages-Waiting: no Voice-Message: 0/0 (0/0)