Hello Everybody! Here are the full SIP logs!
<--- SIP read from 216.48.184.50:5060 ---> INVITE sip:[email protected]:5060;user=phone SIP/2.0 Call-ID: 6998640000475636237-1246542986-18105 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone> Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1 Contact: <sip:[email protected]:5060;user=phone> Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO Supported: timer,100rel Max-Forwards: 70 Content-Length: 255 v=0 o=MG4000|2.0 42386 70624 IN IP4 216.48.184.30 s=- c=IN IP4 216.48.184.30 t=0 0 m=audio 33068 RTP/AVP 18 98 101 13 a=rtpmap:98 G.729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:13 CN/8000 <-------------> [Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (12 headers 12 lines) --- [Jul 2 16:56:26] VERBOSE[13420] logger.c: Sending to 216.48.184.50 : 5060 (no NAT) [Jul 2 16:56:26] VERBOSE[13420] logger.c: Using INVITE request as basis request - 6998640000475636237-1246542986-18105 [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found no matching peer or user for '216.48.184.50:5060' [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 18 [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 98 [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 101 [Jul 2 16:56:26] VERBOSE[13420] logger.c: Found RTP audio format 13 [Jul 2 16:56:26] VERBOSE[13420] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x100 (g729)/video=0x0 (nothing) , combined - 0x0 (nothing) [Jul 2 16:56:26] VERBOSE[13420] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), co mbined - 0x1 (telephone-event) [Jul 2 16:56:26] VERBOSE[13420] logger.c: Looking for 6587972772285297 in didx-to-mor (domain 82.80.231.238) [Jul 2 16:56:26] VERBOSE[13420] logger.c: list_route: hop: <sip:[email protected]:5060;user=phone> [Jul 2 16:56:26] VERBOSE[13420] logger.c: <--- Transmitting (no NAT) to 216.48.184.50:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1;received=216.48.184.50 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone> Call-ID: 6998640000475636237-1246542986-18105 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 <------------> [Jul 2 16:56:26] VERBOSE[7001] logger.c: -- Executing [6587972772285...@didx-to-mor:1] Set("SIP/5060-bc068a30", "CDR(accountcode)=32 9") in new stack [Jul 2 16:56:26] VERBOSE[7001] logger.c: -- Executing [6587972772285...@didx-to-mor:2] Answer("SIP/5060-bc068a30", "") in new stack [Jul 2 16:56:26] VERBOSE[7001] logger.c: Audio is at 82.80.231.238 port 19616 [Jul 2 16:56:26] VERBOSE[7001] logger.c: <--- Reliably Transmitting (no NAT) to 216.48.184.50:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1;received=216.48.184.50 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone>;tag=as2a5d9a27 Call-ID: 6998640000475636237-1246542986-18105 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 150 v=0 o=root 13368 13368 IN IP4 82.80.231.238 s=session c=IN IP4 82.80.231.238 t=0 0 m=audio 19616 RTP/AVP a=silenceSupp:off - - - - a=sendrecv <------------> [Jul 2 16:56:26] VERBOSE[7001] logger.c: -- Executing [6587972772285...@didx-to-mor:3] SayDigits("SIP/5060-bc068a30", "123456789") i n new stack [Jul 2 16:56:26] VERBOSE[7001] logger.c: -- <SIP/5060-bc068a30> Playing 'digits/1' (language 'en') [Jul 2 16:56:26] VERBOSE[13420] logger.c: <--- SIP read from 216.48.184.50:5060 ---> ACK sip:[email protected]:5060;user=phone SIP/2.0 Call-ID: 6998640000475636237-1246542986-18105 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone>;tag=as2a5d9a27 CSeq: 1 ACK Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-1 Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 2 16:56:26] VERBOSE[13420] logger.c: --- (8 headers 0 lines) --- [Jul 2 16:56:27] VERBOSE[13420] logger.c: <--- SIP read from 216.48.184.50:5060 ---> BYE sip:[email protected]:5060;user=phone SIP/2.0 Call-ID: 6998640000475636237-1246542986-18105 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone>;tag=as2a5d9a27 CSeq: 2 BYE Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-2 Supported: timer,100rel Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 2 16:56:27] VERBOSE[13420] logger.c: --- (9 headers 0 lines) --- [Jul 2 16:56:27] VERBOSE[13420] logger.c: Sending to 216.48.184.50 : 5060 (no NAT) [Jul 2 16:56:27] VERBOSE[13420] logger.c: <--- Transmitting (no NAT) to 216.48.184.50:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.48.184.50:5060;branch=z9hG4bK-61202a000330a20d-d830b832-2;received=216.48.184.50 From: <sip:[email protected]:5060;user=phone>;tag=24794 To: <sip:[email protected]:5060;user=phone>;tag=as2a5d9a27 Call-ID: 6998640000475636237-1246542986-18105 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 <------------> Thanks! Elliot _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
