Via: SIP/2.0/UDP 124.7.195.74:5060;branch=z9hG4bK6e15c3f0;rport From: "V0120022047000047493" ;tag=as4589f9ca To: Contact: Call-ID: 02b73ff03182689d19c6e55c00c03080@124.7.195.74 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "V0120022047000047493" ;privacy=off;screen=no Date: Fri, 19 Jan 2007 20:50:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 239 v=0 o=root 31950 31950 IN IP4 124.7.195.74 s=session c=IN IP4 124.7.195.74 t=0 0 m=audio 18458 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 0109#0018289645165@sify localhost*CLI> <-- SIP read from 221.135.102.100:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 124.7.195.74:5060;branch=z9hG4bK6e15c3f0;rport From: "V0120022047000047493" ;tag=as4589f9ca To: Call-ID: 02b73ff03182689d19c6e55c00c03080@124.7.195.74 CSeq: 102 INVITE Content-Length: 0 --- (7 headers 0 lines) --- == Spawn extension (default, 9119047245098, 4) exited non-zero on 'Local/9119047245098@default-02eb,2' -- Executing DeadAGI("Local/9119047245098@default-02eb,2", "call_log|124.7.195.74|localhost") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log -- AGI Script call_log completed, returning 0 -- Executing DeadAGI("Local/9119047245098@default-02eb,2", "new_hangup|124.7.195.74|localhost|16|ANSWER") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/new_hangup -- AGI Script new_hangup completed, returning 0 -- Bailing out, timeout on ast_waitfor(chan, 700). Must have got the silence we needed -- Bailing out, timeout on ast_waitfor(chan, 700). Must have got the silence we needed -- Detected person. Threshhold is 2200 and got 520 -- Executing Set("SIP/sify-0a006c78", "CALLFILENAME=talk-V0120022037000049850-20070120-022049") in new stack -- Executing Monitor("SIP/sify-0a006c78", "wav|/var/spool/asterisk/monitor/DONE/20070120/talk-V0120022037000049850-20070120-022049") in new stack -- Executing AGI("SIP/sify-0a006c78", "transfer_agi|124.7.195.74|localhost") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/transfer_agi -- AGI Script transfer_agi completed, returning 0 -- Executing Dial("SIP/sify-0a006c78", "SIP/108") in new stack We're at 10.10.10.3 port 13502 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 11 lines Reliably Transmitting (NAT) to 10.10.10.115:61940: INVITE sip:108@10.10.10.115:61940;rinstance=26e4a0978eac81f8 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK180a9087;rport From: "V0120022037000049850" ;tag=as13d3caf0 To: Contact: Call-ID: 5036334f58c566010d0bdf8174afb098@10.10.10.3 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "V0120022037000049850" ;privacy=off;screen=no Date: Fri, 19 Jan 2007 20:50:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235 v=0 o=root 31950 31950 IN IP4 10.10.10.3 s=session c=IN IP4 10.10.10.3 t=0 0 m=audio 13502 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 108 localhost*CLI> ex <-- SIP read from 10.10.10.115:61940: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK180a9087;rport=5060 Contact: To: ;tag=d0233431 From: "V0120022037000049850";tag=as13d3caf0 Call-ID: 5036334f58c566010d0bdf8174afb098@10.10.10.3 CSeq: 102 INVITE User-Agent: eyeBeam release 1003l stamp 30937 Content-Length: 0 --- (9 headers 0 lines) --- -- SIP/108-0a19a078 is ringing localhost*CLI> ex <-- SIP read from 10.10.10.115:61940: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK180a9087;rport=5060 Contact: To: ;tag=d0233431 From: "V0120022037000049850";tag=as13d3caf0 Call-ID: 5036334f58c566010d0bdf8174afb098@10.10.10.3 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003l stamp 30937 Content-Length: 259 v=0 o=- 0 2 IN IP4 10.10.10.115 s=CounterPath eyeBeam 1.5 c=IN IP4 10.10.10.115 t=0 0 m=audio 10906 RTP/AVP 18 101 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:B6F6A5D46EEE479ABB2F6DEDF2A9895F --- (11 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.115:10906 Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) 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 10.10.10.115, port 61940 Transmitting (NAT) to 10.10.10.115:61940: ACK sip:108@10.10.10.115:61940;rinstance=26e4a0978eac81f8 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.3:5060;branch=z9hG4bK60c34295;rport From: "V0120022037000049850" ;tag=as13d3caf0 To: ;tag=d0233431 Contact: Call-ID: 5036334f58c566010d0bdf8174afb098@10.10.10.3 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "V0120022037000049850" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/108-0a19a078 answered SIP/sify-0a006c78 localhost*CLI> exit <-- SIP read from 221.135.102.100:5060: SIP/2.0 183 Session Progress To: ;tag=3378228636-322819 From: "V0120022047000047493" ;tag=as4589f9ca Contact: Call-ID: 02b73ff03182689d19c6e55c00c03080@124.7.195.74 CSeq: 102 INVITE Content-Type: application/sdp Via: SIP/2.0/UDP 124.7.195.74:5060;branch=z9hG4bK6e15c3f0;rport Content-Length: 222 v=0 o=nextone-msw1 1234 0 IN IP4 221.135.102.100 s=sip call c=IN IP4 221.135.102.101 t=0 0 m=audio 29602 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no --- (9 headers 10 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 221.135.102.101:29602 Found description format telephone-event Found description format G729 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/sify-09fb2400 is making progress passing it to Local/9118289645165@default-d452,2