ast1*CLI> -- Accepting AUTHENTICATED call from 172.16.64.10: > requested format = alaw, > requested prefs = (alaw|g729), > actual format = g729, > host prefs = (g729|alaw), > priority = mine -- Executing [7469@sip:1] Dial("IAX2/edin-test-2460", "SIP/7469@proxy1") in new stack == Using SIP RTP CoS mark 5 Audio is at 172.16.64.2 port 17338 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.16.64.5:5060: INVITE sip:7469@172.16.64.5 SIP/2.0 Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3d1c6d60;rport Max-Forwards: 70 From: "8888" ;tag=as34002c7d To: Contact: Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.1 Date: Wed, 17 Jun 2009 08:20:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 v=0 o=root 198748345 198748345 IN IP4 172.16.64.2 s=Asterisk PBX 1.6.1.1 c=IN IP4 172.16.64.2 t=0 0 m=audio 17338 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 7469@proxy1 ast1*CLI> <--- SIP read from UDP://172.16.64.5:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3d1c6d60;rport=5060 From: "8888" ;tag=as34002c7d To: Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 CSeq: 102 INVITE Server: OpenSIPS (1.5.1-notls (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- ast1*CLI> <--- SIP read from UDP://172.16.64.5:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.64.2:5060;received=172.16.64.2;branch=z9hG4bK3d1c6d60;rport=5060 From: "8888" ;tag=as34002c7d To: ;tag=001906af068d006645f9621c-b5273c88 Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 Date: Wed, 17 Jun 2009 08:20:32 GMT CSeq: 102 INVITE Server: Cisco-CP7911G/8.5.2 Contact: Record-Route: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "7469" ;party=called;id-type=subscriber;privacy=off;screen=yes Allow-Events: kpml,dialog Content-Length: 0 <-------------> --- (14 headers 0 lines) --- -- SIP/proxy1-00f26568 is ringing Scheduling destruction of SIP dialog '3b1e2b495732b46765d269616e322f13@172.16.64.5' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 172.16.64.5:5060: CANCEL sip:7469@172.16.64.5 SIP/2.0 Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3d1c6d60;rport Max-Forwards: 70 From: "8888" ;tag=as34002c7d To: Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.1.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '3b1e2b495732b46765d269616e322f13@172.16.64.5' in 32000 ms (Method: INVITE) == Spawn extension (sip, 7469, 1) exited non-zero on 'IAX2/edin-test-2460' -- Hungup 'IAX2/edin-test-2460' ast1*CLI> <--- SIP read from UDP://172.16.64.5:5060 ---> SIP/2.0 200 canceling Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3d1c6d60;rport=5060 From: "8888" ;tag=as34002c7d To: ;tag=e3d3f0831f931c5fcb7930ce92d15b13-f079 Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 CSeq: 102 CANCEL Server: OpenSIPS (1.5.1-notls (x86_64/linux)) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- ast1*CLI> <--- SIP read from UDP://172.16.64.5:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 172.16.64.2:5060;received=172.16.64.2;branch=z9hG4bK3d1c6d60;rport=5060 From: "8888" ;tag=as34002c7d To: ;tag=001906af068d006645f9621c-b5273c88 Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 Date: Wed, 17 Jun 2009 08:20:35 GMT CSeq: 102 INVITE Server: Cisco-CP7911G/8.5.2 Contact: Record-Route: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Remote-Party-ID: "7469" ;party=called;id-type=subscriber;privacy=off;screen=yes Allow-Events: kpml,dialog Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Transmitting (no NAT) to 172.16.64.5:5060: ACK sip:7469@172.16.64.5 SIP/2.0 Via: SIP/2.0/UDP 172.16.64.2:5060;branch=z9hG4bK3d1c6d60;rport Max-Forwards: 70 From: "8888" ;tag=as34002c7d To: ;tag=001906af068d006645f9621c-b5273c88 Contact: Call-ID: 3b1e2b495732b46765d269616e322f13@172.16.64.5 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.1 Content-Length: 0 --- Really destroying SIP dialog '3b1e2b495732b46765d269616e322f13@172.16.64.5' Method: INVITE