User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 28 Jun 2007 17:44:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 284 v=0 o=root 4297 4297 IN IP4 91.103.25.29 s=session c=IN IP4 91.103.25.29 t=0 0 m=audio 12670 RTP/AVP 4 18 3 101 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Transmitting (no NAT) to 80.93.57.101:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101 From: ;tag=ffffffff1d1a3c1 To: ;tag=as6bbe00cb Call-ID: ffffffff1d1a3c1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk*CLI> <-- SIP read from 91.103.25.26:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- <-- SIP read from 91.103.25.26:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- asterisk*CLI> <-- SIP read from 91.103.25.26:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: ;tag=31c3ef99 Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- <-- SIP read from 91.103.25.26:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: ;tag=31c3ef99 Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- asterisk*CLI> <-- SIP read from 91.103.25.26:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: ;tag=31c3ef99 Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 130 v=0 o=- 8829 8829 IN IP4 192.168.0.222 s=session c=IN IP4 192.168.0.222 t=0 0 m=audio 10006 RTP/AVP 4 a=rtpmap:4 G723/8000 --- (9 headers 7 lines) --- Found RTP audio format 4 Peer audio RTP is at port 192.168.0.222:10006 Found description format G723 Capabilities: us - 0x80103 (g723|gsm|g729|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.222, port 5060 Transmitting (NAT) to 91.103.25.26:5060: ACK sip:22@192.168.0.222:5060 SIP/2.0 Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK012ab263;rport From: "NOC" ;tag=as4c006281 To: ;tag=31c3ef99 Contact: Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 91.103.25.29 port 19006 Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 80.93.57.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101 From: ;tag=ffffffff1d1a3c1 To: ;tag=as6bbe00cb Call-ID: ffffffff1d1a3c1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 4297 4297 IN IP4 91.103.25.29 s=session c=IN IP4 91.103.25.29 t=0 0 m=audio 19006 RTP/AVP 4 96 a=rtpmap:4 G723/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 91.103.25.26:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK40e6962b From: "NOC";tag=as4c006281 To: ;tag=31c3ef99 Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 130 v=0 o=- 8829 8829 IN IP4 192.168.0.222 s=session c=IN IP4 192.168.0.222 t=0 0 m=audio 10006 RTP/AVP 4 a=rtpmap:4 G723/8000 --- (9 headers 7 lines) --- Found RTP audio format 4 Peer audio RTP is at port 192.168.0.222:10006 Found description format G723 Capabilities: us - 0x80103 (g723|gsm|g729|h263), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.222, port 5060 Transmitting (NAT) to 91.103.25.26:5060: ACK sip:22@192.168.0.222:5060 SIP/2.0 Via: SIP/2.0/UDP 91.103.25.29:5060;branch=z9hG4bK012ab263;rport From: "NOC" ;tag=as4c006281 To: ;tag=31c3ef99 Contact: Call-ID: 718127d30126fd87644ed1770a09baa7@91.103.25.29 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- We're at 91.103.25.29 port 19006 Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 80.93.57.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091;received=80.93.57.101 From: ;tag=ffffffff1d1a3c1 To: ;tag=as6bbe00cb Call-ID: ffffffff1d1a3c1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 211 v=0 o=root 4297 4297 IN IP4 91.103.25.29 s=session c=IN IP4 91.103.25.29 t=0 0 m=audio 19006 RTP/AVP 4 96 a=rtpmap:4 G723/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 80.93.57.101:5060: ACK sip:577266@91.103.25.29 SIP/2.0 Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091 From: ;tag=ffffffff1d1a3c1 To: ;tag=as6bbe00cb Call-ID: ffffffff1d1a3c1 Contact: CSeq: 1 ACK Accept: application/sdp Max-Forwards: 70 User-Agent: IXC SS Content-Length: 0 --- (11 headers 0 lines) --- <-- SIP read from 80.93.57.101:5060: ACK sip:577266@91.103.25.29 SIP/2.0 Via: SIP/2.0/UDP 80.93.57.101;branch=z9hG4bK.1091 From: ;tag=ffffffff1d1a3c1 To: ;tag=as6bbe00cb Call-ID: ffffffff1d1a3c1 Contact: CSeq: 1 ACK Accept: application/sdp Max-Forwards: 70 User-Agent: IXC SS Content-Length: 0