Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex| ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone -event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.105:8000 Looking for 123 in mycontext (domain 192.168.1.102) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.105:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7 8B;received=192.168.1.105;rport=5060 From: me ;tag=2509996786 To: Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [123@mycontext:1] Dial("SIP/me-0988e160", "SIP/123@home|50") in new stack Audio is at 88.149.162.157 port 17416 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 ate: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 123@home Retransmitting #1 (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (NAT) to 81.174.8.170:5060: INVITE sip:123@81.174.8.170 SIP/2.0 Via: SIP/2.0/UDP 88.149.162.157:5060;branch=z9hG4bK1065626f;rport From: "me" ;tag=as14f08839 To: Contact: Call-ID: 391f9b8e4f33949979aeef8c0d945de3@88.149.162.157 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 31 Jan 2007 15:51:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 241 v=0 o=root 9339 9339 IN IP4 88.149.162.157 s=session c=IN IP4 88.149.162.157 t=0 0 m=audio 17416 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- rfc-1918*CLI> <--- SIP read from 192.168.1.105:5060 ---> CANCEL sip:123@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C 12A4D78B From: me ;tag=2509996786 To: Contact: Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 CANCEL Proxy-Authorization: Digest username="me",realm="asterisk",nonce="374691c0",resp onse="71cb3b9b2e18a88c0641b44709329d77",uri="sip:123@192.168.1.102",algorithm=MD 5 Max-Forwards: 70 User-Agent: X-Lite release 1103a Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.1.105 : 5060 (NAT) <--- Reliably Transmitting (NAT) to 192.168.1.105:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7 8B;received=192.168.1.105;rport=5060 From: me ;tag=2509996786 To: ;tag=as49d906ae Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 192.168.1.105:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bKDB7566B661F04741ACC7734C12A4D7 8B;received=192.168.1.105;rport=5060 From: me ;tag=2509996786 To: ;tag=as49d906ae Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (mycontext, 123, 1) exited non-zero on 'SIP/me-0988e160' rfc-1918*CLI> <--- SIP read from 192.168.1.105:5060 ---> ACK sip:123@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C 12A4D78B From: me ;tag=2509996786 To: ;tag=as49d906ae Contact: Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 ' Method: ACK <--- SIP read from 192.168.1.105:5060 ---> ACK sip:123@192.168.1.102 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105:5060;rport;branch=z9hG4bKDB7566B661F04741ACC7734C 12A4D78B From: me ;tag=2509996786 To: ;tag=as49d906ae Contact: Call-ID: 9336F145-18AF-41D1-8407-6ED86E4EFB52@192.168.1.105 CSeq: 15527 ACK Max-Forwards: 70 Content-Length: 0 rfc-1918*CLI> <-------------> --- (9 headers 0 lines) --- rfc-1918*CLI>