-- Executing [870600719795@satdirect-dial:1] Dial("IAX2/jnctn-outbound-5", "SIP/870600719795@telnor") in new stack Audio is at 216.219.244.206 port 16704 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 63.110.124.53:5060: INVITE sip:870600719795@63.110.124.53 SIP/2.0 Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK15f112e1;rport Max-Forwards: 70 From: "19547072498" ;tag=as2ebb4a5c To: Contact: Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 14 Nov 2006 03:24:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 246 v=0 o=root 22368 22368 IN IP4 216.219.244.206 s=session c=IN IP4 216.219.244.206 t=0 0 m=audio 16704 RTP/AVP 0 101 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 870600719795@telnor <--- SIP read from 63.110.124.53:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK15f112e1;rport From: "19547072498" ;tag=as2ebb4a5c To: Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from 63.110.124.53:5060 ---> SIP/2.0 180 Ringing To: ;tag=3372463698-975858 From: "19547072498" ;tag=as2ebb4a5c Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 102 INVITE Contact: sip:870600719795@63.110.124.53:5060 Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK15f112e1;rport Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/telnor-085a8e80 is ringing <--- SIP read from 63.110.124.53:5060 ---> SIP/2.0 200 OK To: ;tag=3372463698-975858 From: "19547072498" ;tag=as2ebb4a5c Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 102 INVITE Contact: sip:870600719795@63.110.124.53:5060 Content-Type: application/sdp Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK15f112e1;rport Content-Length: 227 v=0 o=shbysbc2 0 0 IN IP4 63.110.124.53 s=sip call c=IN IP4 63.110.124.49 t=0 0 m=audio 10930 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 <-------------> --- (9 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 63.110.124.49:10930 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 63.110.124.49:10930 --- set_address_from_contact host '63.110.124.53' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 63.110.124.53, port 5060 Transmitting (no NAT) to 63.110.124.53:5060: ACK sip:870600719795@63.110.124.53:5060 SIP/2.0 Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK4f41fe7b;rport Max-Forwards: 70 From: "19547072498" ;tag=as2ebb4a5c To: ;tag=3372463698-975858 Contact: Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/telnor-085a8e80 answered IAX2/jnctn-outbound-5 <--- SIP read from 63.110.124.53:5060 ---> INVITE sip:19547072498@216.219.244.206 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: "19547072498" ;tag=as2ebb4a5c From: ;tag=3372463698-975858 Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 2 INVITE Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bK43098c86274cb0f234900a29d5c87151 Contact: sip:870600719795@63.110.124.53:5060 Content-Type: application/sdp Content-Length: 348 v=0 o=shbysbc2 0 1 IN IP4 63.110.124.53 s=sip call c=IN IP4 63.110.124.49 t=0 0 m=image 10930 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (12 headers 15 lines) --- Sending to 63.110.124.53 : 5060 (no NAT) Got T.38 offer in SDP in dialog 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 Capabilities: us - 0x4 (ulaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) <--- Reliably Transmitting (no NAT) to 63.110.124.53:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bK43098c86274cb0f234900a29d5c87151;received=63.110.124.53 From: ;tag=3372463698-975858 To: "19547072498" ;tag=as2ebb4a5c Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> <--- SIP read from 63.110.124.53:5060 ---> ACK sip:19547072498@216.219.244.206 SIP/2.0 Max-Forwards: 69 From: ;tag=3372463698-975858 To: "19547072498" ;tag=as2ebb4a5c Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 2 ACK Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bK43098c86274cb0f234900a29d5c87151 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 63.110.124.53:5060 ---> INVITE sip:19547072498@216.219.244.206 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: "19547072498" ;tag=as2ebb4a5c From: ;tag=3372463698-975858 Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 3 INVITE Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bK02b03500f018719dca02502d9142bd86 Contact: sip:870600719795@63.110.124.53:5060 Content-Type: application/sdp Content-Length: 159 v=0 o=shbysbc2 0 2 IN IP4 63.110.124.53 s=sip call c=IN IP4 63.110.124.49 t=0 0 m=audio 10930 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - <-------------> --- (12 headers 8 lines) --- Sending to 63.110.124.53 : 5060 (no NAT) Found RTP audio format 0 Peer audio RTP is at port 63.110.124.49:10930 Found description format PCMU for ID 0 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 63.110.124.49:10930 Audio is at 216.219.244.206 port 16704 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 63.110.124.53:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bK02b03500f018719dca02502d9142bd86;received=63.110.124.53 From: ;tag=3372463698-975858 To: "19547072498" ;tag=as2ebb4a5c Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 190 v=0 o=root 22368 22369 IN IP4 216.219.244.206 s=session c=IN IP4 216.219.244.206 t=0 0 m=audio 16704 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 63.110.124.53:5060 ---> ACK sip:19547072498@216.219.244.206 SIP/2.0 Max-Forwards: 69 To: "19547072498" ;tag=as2ebb4a5c From: ;tag=3372463698-975858 Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 3 ACK Via: SIP/2.0/UDP 63.110.124.53:5060;branch=z9hG4bKb00e54577188e2b7cdc80ad061699c77 Contact: sip:870600719795@63.110.124.53:5060 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- *** *** *** *** *** I hangup here since there is no audio *** *** *** *** *CLI> Scheduling destruction of SIP dialog '3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 63.110.124.53, port 5060 Reliably Transmitting (no NAT) to 63.110.124.53:5060: BYE sip:870600719795@63.110.124.53:5060 SIP/2.0 Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK130837f4;rport Max-Forwards: 70 From: "19547072498" ;tag=as2ebb4a5c To: ;tag=3372463698-975858 Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- == Spawn extension (satdirect-dial, 870600719795, 1) exited non-zero on 'IAX2/jnctn-outbound-5' -- Executing [h@satdirect-dial:1] Hangup("IAX2/jnctn-outbound-5", "") in new stack == Spawn extension (satdirect-dial, h, 1) exited non-zero on 'IAX2/jnctn-outbound-5' -- Hungup 'IAX2/jnctn-outbound-5' <--- SIP read from 63.110.124.53:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.219.244.206:5060;branch=z9hG4bK130837f4;rport From: "19547072498" ;tag=as2ebb4a5c To: ;tag=3372463698-975858 Call-ID: 3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206 CSeq: 103 BYE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '3edf0b9e09a2d3630ece68eb04fb6e92@216.219.244.206' Method: ACK