root@improvise:/etc/asterisk# asterisk -r Asterisk 1.4.23.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 1.4.23.2 currently running on improvise (pid = 6586) Verbosity is at least 10 Core debug is at least 10 -- Remote UNIX connection improvise*CLI> sip set debug SIP Debugging enabled improvise*CLI> core set verbose 10 Verbosity is at least 10 improvise*CLI> improvise*CLI> -- Starting simple switch on 'Zap/4-1' -- Executing [s@from-pstn:1] Dial("Zap/4-1", "SIP/phone@phone|10") in new stack Audio is at 192.168.1.106 port 17720 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.106:5060: INVITE sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;rport From: "Cell Phone VA" ;tag=as43af87c5 To: Contact: Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 15 Mar 2009 22:19:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 6586 6586 IN IP4 192.168.1.106 s=session c=IN IP4 192.168.1.106 t=0 0 m=audio 17720 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called phone@phone improvise*CLI> <--- SIP read from 192.168.1.106:5060 ---> INVITE sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;rport From: "Cell Phone VA" ;tag=as43af87c5 To: Contact: Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 15 Mar 2009 22:19:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 6586 6586 IN IP4 192.168.1.106 s=session c=IN IP4 192.168.1.106 t=0 0 m=audio 17720 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- improvise*CLI> <--- Transmitting (no NAT) to 192.168.1.106:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;received=192.168.1.106;rport=5060 From: "Cell Phone VA" ;tag=as43af87c5 To: ;tag=as43af87c5 Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '58d00970079c8700075185b64fc6f37f@192.168.1.106' in 32000 ms (Method: INVITE) improvise*CLI> <--- SIP read from 192.168.1.106:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;received=192.168.1.106;rport=5060 From: "Cell Phone VA" ;tag=as43af87c5 To: ;tag=as43af87c5 Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Got SIP response 482 "Loop Detected" back from 192.168.1.106 Transmitting (no NAT) to 192.168.1.106:5060: ACK sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;rport From: "Cell Phone VA" ;tag=as43af87c5 To: ;tag=as43af87c5 Contact: Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- improvise*CLI> <--- SIP read from 192.168.1.106:5060 ---> ACK sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK1a0070f4;rport From: "Cell Phone VA" ;tag=as43af87c5 To: ;tag=as43af87c5 Contact: Call-ID: 58d00970079c8700075185b64fc6f37f@192.168.1.106 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Now forwarding Zap/4-1 to 'Local/phone@phone1' (thanks to SIP/phone-09c3ca90) [Mar 15 18:19:43] NOTICE[32059]: chan_local.c:498 local_call: No such extension/context phone@phone1 while calling Local channel [Mar 15 18:19:43] NOTICE[32059]: app_dial.c:554 wait_for_answer: Failed to dial on local channel for call forward to 'Local' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@from-pstn:2] VoiceMail("Zap/4-1", "line") in new stack [Mar 15 18:19:43] WARNING[32059]: app_voicemail.c:3896 leave_voicemail: No entry in voicemail config file for 'line' -- Executing [s@from-pstn:3] Hangup("Zap/4-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Really destroying SIP dialog '58d00970079c8700075185b64fc6f37f@192.168.1.106' Method: ACK -- Starting simple switch on 'Zap/4-1' [Mar 15 18:19:56] WARNING[32085]: chan_dahdi.c:6665 ss_thread: CallerID returned with error on channel 'Zap/4-1' -- Executing [s@from-pstn:1] Dial("Zap/4-1", "SIP/phone@phone|10") in new stack Audio is at 192.168.1.106 port 11398 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.106:5060: INVITE sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;rport From: "asterisk" ;tag=as28df5b9d To: Contact: Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 15 Mar 2009 22:19:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 6586 6586 IN IP4 192.168.1.106 s=session c=IN IP4 192.168.1.106 t=0 0 m=audio 11398 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- improvise*CLI> <--- SIP read from 192.168.1.106:5060 ---> INVITE sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;rport From: "asterisk" ;tag=as28df5b9d To: Contact: Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 15 Mar 2009 22:19:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 6586 6586 IN IP4 192.168.1.106 s=session c=IN IP4 192.168.1.106 t=0 0 m=audio 11398 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 14 lines) --- -- Called phone@phone improvise*CLI> <--- Transmitting (no NAT) to 192.168.1.106:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;received=192.168.1.106;rport=5060 From: "asterisk" ;tag=as28df5b9d To: ;tag=as28df5b9d Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6fd34d0944597c7d742cd8b8306d1272@192.168.1.106' in 32000 ms (Method: INVITE) <--- SIP read from 192.168.1.106:5060 ---> SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;received=192.168.1.106;rport=5060 From: "asterisk" ;tag=as28df5b9d To: ;tag=as28df5b9d Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Got SIP response 482 "Loop Detected" back from 192.168.1.106 Transmitting (no NAT) to 192.168.1.106:5060: ACK sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;rport From: "asterisk" ;tag=as28df5b9d To: ;tag=as28df5b9d Contact: Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 192.168.1.106:5060 ---> ACK sip:phone@192.168.1.106 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5060;branch=z9hG4bK10071973;rport From: "asterisk" ;tag=as28df5b9d To: ;tag=as28df5b9d Contact: Call-ID: 6fd34d0944597c7d742cd8b8306d1272@192.168.1.106 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Now forwarding Zap/4-1 to 'Local/phone@phone1' (thanks to SIP/phone-09c3c6a0) [Mar 15 18:19:56] NOTICE[32085]: chan_local.c:498 local_call: No such extension/context phone@phone1 while calling Local channel [Mar 15 18:19:56] NOTICE[32085]: app_dial.c:554 wait_for_answer: Failed to dial on local channel for call forward to 'Local' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@from-pstn:2] VoiceMail("Zap/4-1", "line") in new stack [Mar 15 18:19:56] WARNING[32085]: app_voicemail.c:3896 leave_voicemail: No entry in voicemail config file for 'line' -- Executing [s@from-pstn:3] Hangup("Zap/4-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' Really destroying SIP dialog '6fd34d0944597c7d742cd8b8306d1272@192.168.1.106' Method: ACK