SIP Debugging enabled [Jul 26 20:02:04] -- Accepting AUTHENTICATED call from 70.103.254.3: > requested format = ulaw, > requested prefs = (ulaw), > actual format = ulaw, > host prefs = (ulaw), > priority = mine [Jul 26 20:02:04] -- Executing Set("IAX2/voip1-1", "ORIGINAL_EXT=16024897001") in new stack [Jul 26 20:02:04] -- Executing GotoIf("IAX2/voip1-1", "1?clients:dial_out") in new stack [Jul 26 20:02:04] -- Goto (OUTBOUND,16024897001,3) [Jul 26 20:02:04] -- Executing Goto("IAX2/voip1-1", "LKR-CLIENTS|6024897001|1") in new stack [Jul 26 20:02:04] -- Goto (LKR-CLIENTS,6024897001,1) [Jul 26 20:02:04] -- Sent into invalid extension '6024897001' in context 'LKR-CLIENTS' on IAX2/voip1-1 [Jul 26 20:02:04] -- Executing Goto("IAX2/voip1-1", "OUTBOUND|16024897001|dial_out") in new stack [Jul 26 20:02:04] -- Goto (OUTBOUND,16024897001,4) [Jul 26 20:02:04] -- Executing Goto("IAX2/voip1-1", "DIAL_OUT|16024897001|1") in new stack [Jul 26 20:02:04] -- Goto (DIAL_OUT,16024897001,1) [Jul 26 20:02:04] -- Executing Dial("IAX2/voip1-1", "Sip/69114416024897001@telcommone") in new stack [Jul 26 20:02:04] We're at 70.103.254.2 port 30548 [Jul 26 20:02:04] Adding codec 0x4 (ulaw) to SDP [Jul 26 20:02:04] Adding non-codec 0x1 (telephone-event) to SDP [Jul 26 20:02:04] 13 headers, 10 lines [Jul 26 20:02:04] Reliably Transmitting (no NAT) to 72.232.115.250:5060: INVITE sip:69114416024897001@sip1.telcommone.net SIP/2.0 Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;rport From: "Tim Ferguson" ;tag=as36b2ff08 To: Contact: Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 27 Jul 2006 03:02:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 3868 3868 IN IP4 70.103.254.2 s=session c=IN IP4 70.103.254.2 t=0 0 m=audio 30548 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Jul 26 20:02:04] -- Called 69114416024897001@telcommone [Jul 26 20:02:04] <-- SIP read from 72.232.115.250:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;received=70.103.254.2;rport=5060 From: "Tim Ferguson" ;tag=as36b2ff08 To: Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 102 INVITE User-Agent: TelCommOne-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 [Jul 26 20:02:04] --- (10 headers 0 lines)[Jul 26 20:02:04] --- [Jul 26 20:02:07] <-- SIP read from 72.232.115.250:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;received=70.103.254.2;rport=5060 From: "Tim Ferguson" ;tag=as36b2ff08 To: ;tag=as3a4b1f47 Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 102 INVITE User-Agent: TelCommOne-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 2486 2486 IN IP4 72.232.115.250 s=session c=IN IP4 72.232.115.250 t=0 0 m=audio 19230 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - [Jul 26 20:02:07] --- (11 headers 10 lines)[Jul 26 20:02:07] --- [Jul 26 20:02:07] Found RTP audio format 0 [Jul 26 20:02:07] Found RTP audio format 101 [Jul 26 20:02:07] Peer audio RTP is at port 72.232.115.250:19230 [Jul 26 20:02:07] Found description format PCMU [Jul 26 20:02:07] Found description format telephone-event [Jul 26 20:02:07] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jul 26 20:02:07] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 26 20:02:07] list_route: hop: [Jul 26 20:02:07] set_destination: Parsing for address/port to send to [Jul 26 20:02:07] set_destination: set destination to 72.232.115.250, port 5060 [Jul 26 20:02:07] Transmitting (no NAT) to 72.232.115.250:5060: ACK sip:69114416024897001@72.232.115.250 SIP/2.0 Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK44ed5f7a;rport From: "Tim Ferguson" ;tag=as36b2ff08 To: ;tag=as3a4b1f47 Contact: Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jul 26 20:02:07] -- SIP/telcommone-09e43458 answered IAX2/voip1-1 [Jul 26 20:02:10] set_destination: Parsing for address/port to send to [Jul 26 20:02:10] set_destination: set destination to 72.232.115.250, port 5060 [Jul 26 20:02:10] Reliably Transmitting (no NAT) to 72.232.115.250:5060: BYE sip:69114416024897001@72.232.115.250 SIP/2.0 Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK3b843a66;rport From: "Tim Ferguson" ;tag=as36b2ff08 To: ;tag=as3a4b1f47 Contact: Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jul 26 20:02:10] == Spawn extension (DIAL_OUT, 16024897001, 1) exited non-zero on 'IAX2/voip1-1' [Jul 26 20:02:10] -- Hungup 'IAX2/voip1-1' [Jul 26 20:02:10] <-- SIP read from 72.232.115.250:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK3b843a66;received=70.103.254.2;rport=5060 From: "Tim Ferguson" ;tag=as36b2ff08 To: ;tag=as3a4b1f47 Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2 CSeq: 103 BYE User-Agent: TelCommOne-GW Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing [Jul 26 20:02:10] --- (11 headers 0 lines)[Jul 26 20:02:10] --- [Jul 26 20:02:10] Destroying call '3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2' teamics0101*CLI> sip no debug