atomical*CLI> -- Attempting call on SIP/18155551212@sip.inphonex.com for s@outgoing:1 (Retry 1) We're at 64.34.205.41 port 13958 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 208.239.76.163:5060: INVITE sip:18155551212@sip.inphonex.com SIP/2.0 Via: SIP/2.0/UDP 64.34.205.41:5060;branch=z9hG4bK0b0da4c1;rport From: "asterisk" ;tag=as6bd12ed4 To: Contact: Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 28 Mar 2007 04:45:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 13591 13591 IN IP4 64.34.205.41 s=session c=IN IP4 64.34.205.41 t=0 0 m=audio 13958 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- atomical*CLI> <-- SIP read from 208.239.76.163:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.34.205.41:5060;received=64.34.205.41;branch=z9hG4bK0b0da4c1;rport=5060 From: "asterisk" ;tag=as6bd12ed4 To: Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 INVITE --- (6 headers 0 lines) --- atomical*CLI> <-- SIP read from 208.239.76.163:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.34.205.41:5060;received=64.34.205.41;branch=z9hG4bK0b0da4c1;rport=5060 From: "asterisk" ;tag=as6bd12ed4 To: ;tag=SDbasfb99-as2eefa867 Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 INVITE User-Agent: IPX Softswitch 2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 267 v=0 o=root 27288 27288 IN IP4 208.239.76.163 s=session c=IN IP4 208.239.76.163 t=0 0 m=audio 25758 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 12 lines) --- Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 208.239.76.163:25758 Found description format GSM Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) 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 208.239.76.163, port 5060 Transmitting (no NAT) to 208.239.76.163:5060: ACK sip:18155551212@208.239.76.163:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.34.205.41:5060;branch=z9hG4bK493dc3a7;rport From: "asterisk" ;tag=as6bd12ed4 To: ;tag=SDbasfb99-as2eefa867 Contact: Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- > Channel SIP/sip.inphonex.com-0818baa8 was answered. -- Executing Monitor("SIP/sip.inphonex.com-0818baa8", "wav|mysound") in new stack -- Executing Wait("SIP/sip.inphonex.com-0818baa8", "15") in new stack atomical*CLI> <-- SIP read from 208.239.76.163:5060: BYE sip:asterisk@64.34.205.41 SIP/2.0 Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bK0gbb3b10fgc0qgs3v300sd00000j1.1 From: ;tag=SDbasfb99-as2eefa867 To: "asterisk" ;tag=as6bd12ed4 Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 BYE User-Agent: IPX Softswitch 2.0 Max-Forwards: 69 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 208.239.76.163 : 5060 (non-NAT) Transmitting (no NAT) to 208.239.76.163:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bK0gbb3b10fgc0qgs3v300sd00000j1.1;received=208.239.76.163 From: ;tag=SDbasfb99-as2eefa867 To: "asterisk" ;tag=as6bd12ed4 Call-ID: 21afa60f1b21ecba21b590a428187d2e@inphonex.com CSeq: 102 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (outgoing, s, 2) exited non-zero on 'SIP/sip.inphonex.com-0818baa8' Mar 28 04:45:38 NOTICE[15687]: pbx_spool.c:279 attempt_thread: Call completed to SIP/18155551212@sip.inphonex.com Destroying call '21afa60f1b21ecba21b590a428187d2e@inphonex.com'