--- (12 headers 0 lines) --- Sending to 192.168.0.50 : 12714 (NAT) Transmitting (NAT) to 192.168.0.50:12714: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:12714 ;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714 From: "cc106"<sip:[email protected] <sip%[email protected]>>;tag=7f1cff22 To: "817275691533"<sip:[email protected]<sip%[email protected]> >;tag=as02559696 Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU. CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected] <sip%[email protected]>> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing
--- Scheduling destruction of call '[email protected]' in 32000 ms set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to set_destination: set destination to 8.14.xxx.xxx, port 5060 Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060: BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport From: "cc106" <sip:[email protected]>;tag=as3f9466a7 To: <sip:[email protected]>;tag=1902000923108720995156225 Call-ID: [email protected] CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc106-b7a1a9d0' -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)") in new stack -- AGI Script agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12) completed, returning 0 Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.' vicidialnow*CLI> <-- SIP read from 8.14.xxx.xxx:5060: SIP/2.0 200 OK CSeq: 102 INVITE Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport From: "V0219160007000134649" <sip:[email protected]>;tag=as79fae976 Call-ID: [email protected] To: <sip:[email protected]>;tag=1902000923098720982816221 Contact: <sip:8.14.xxx.xxx:5060;transport=udp> Content-Type: application/sdp Content-Length: 225 v=0 o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx s=VoipSIP i=Audio Session c=IN IP4 8.14.xxx.xxx t=0 0 m=audio 6220 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (9 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 8.14.xxx.xxx:6220 Found description format G729 Found description format telephone-event Capabilities: us - 0x104 (ulaw|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: <sip:8.14.xxx.xxx:5060;transport=udp> set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to set_destination: set destination to 8.14.xxx.xxx, port 5060 Transmitting (no NAT) to 8.14.xxx.xxx:5060: ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport From: "V0219160007000134649" <sip:[email protected]>;tag=as79fae976 To: <sip:[email protected]>;tag=1902000923098720982816221 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0
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