Use 'exit' when done Asterisk 1.4.2, Copyright (C) 1999 - 2006 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found  == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.2 currently running on hfemsrv (pid = 18420) hfemsrv*CLI> Verbosity is at least 5 hfemsrv*CLI> sip debug hfemsrv*CLI> SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. hfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060: OPTIONS sip:192.168.45.129 SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK2508d83c;rport From: "asterisk" ;tag=as2cc96e52 To: Contact: Call-ID: 3ee92dbe77f51a1748f736be4593719d@161.49.142.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Apr 2007 19:25:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 200 OK From: "asterisk";tag=as2cc96e52 To: ;tag=812da8c0-13c4-46277c06-279cd106-42ff Call-ID: 3ee92dbe77f51a1748f736be4593719d@161.49.142.250 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK2508d83c Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- hfemsrv*CLI> Really destroying SIP dialog '3ee92dbe77f51a1748f736be4593719d@161.49.142.250' Method: OPTIONS hfemsrv*CLI> -- Attempting call on SIP/QuadNortel/7113 for smvoice_callprogress@smvoice-dialout:1 (Retry 1) hfemsrv*CLI> Audio is at 161.49.142.250 port 10000 hfemsrv*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP hfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060: INVITE sip:7113@192.168.45.129 SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK11268a7d;rport From: "Admin System 34" ;tag=as4e5a553d To: Contact: Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 19 Apr 2007 19:25:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 212 v=0 o=root 18420 18420 IN IP4 161.49.142.250 s=session c=IN IP4 161.49.142.250 t=0 0 m=audio 10000 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 100 Trying From: "Admin System 34";tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 102 INVITE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 180 Ringing From: "Admin System 34";tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 102 INVITE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 200 OK From: "Admin System 34";tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 102 INVITE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK11268a7d Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 P-Asserted-Identity: Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/SDP Content-Length: 137 v=0 o=- 91 1 IN IP4 192.168.45.129 s=- t=0 0 m=audio 5260 RTP/AVP 0 c=IN IP4 192.168.45.199 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (14 headers 9 lines) --- hfemsrv*CLI> Found RTP audio format 0 Peer audio RTP is at port 192.168.45.199:5260 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.45.199:5260 hfemsrv*CLI> [Apr 19 14:26:03] WARNING[18442]: chan_sip.c:7724 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '7113' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.45.129, port 5060 Transmitting (no NAT) to 192.168.45.129:5060: ACK sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK4245bbdd;rport From: "Admin System 34" ;tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Contact: Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.45.129, port 5060 hfemsrv*CLI> Reliably Transmitting (no NAT) to 192.168.45.129:5060: BYE sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK1be97cfa;rport From: "Admin System 34" ;tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- Scheduling destruction of SIP dialog '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' in 6400 ms (Method: INVITE) > Channel SIP/QuadNortel-09a4c0e0 was answered. hfemsrv*CLI> -- Executing [smvoice_callprogress@smvoice-dialout:1] GotoIf("SIP/QuadNortel-09a4c0e0", "1?smvoice_callprogress|3:smvoice_callprogress|2") in new stack -- Goto (smvoice-dialout,smvoice_callprogress,3) -- Executing [smvoice_callprogress@smvoice-dialout:3] AGI("SIP/QuadNortel-09a4c0e0", "smvoice|-digium_asterisk") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice hfemsrv*CLI> <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 200 OK From: "Admin System 34";tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 103 BYE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK1be97cfa Supported: 100rel,sipvc,replaces User-Agent: Nortel CS1000 SIP GW release_4.0 version_sse-4.00.55 Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- hfemsrv*CLI> -- Playing '/tmp/smvoice.19294_0' (escape_digits=0123456789#) (sample_offset 0) hfemsrv*CLI> quit -- Playing '/tmp/smvoice.19294_0' (escape_digits=0123456789#) (sample_offset 0) hfemsrv*CLI> quit [Apr 19 14:26:09] WARNING[18442]: chan_sip.c:2013 __sip_autodestruct: Autodestruct on dialog '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' with owner in place (Method: INVITE) hfemsrv*CLI> quit == Spawn extension (smvoice-dialout, smvoice_callprogress, 3) exited non-zero on 'SIP/QuadNortel-09a4c0e0' hfemsrv*CLI> quit Scheduling destruction of SIP dialog '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' in 6400 ms (Method: INVITE) hfemsrv*CLI> quit set_destination: Parsing for address/port to send to hfemsrv*CLI> quit set_destination: set destination to 192.168.45.129, port 5060 hfemsrv*CLI> quit Reliably Transmitting (no NAT) to 192.168.45.129:5060: BYE sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 161.49.142.250:5060;branch=z9hG4bK4ede3adc;rport From: "Admin System 34" ;tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- hfemsrv*CLI> quit [Apr 19 14:26:09] NOTICE[19253]: pbx_spool.c:351 attempt_thread: Call completed to SIP/QuadNortel/7113 hfemsrv*CLI> quit <--- SIP read from 192.168.45.129:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist From: "Admin System 34";tag=as4e5a553d To: ;tag=812da8c0-13c4-46277c0a-279ce16e-4a2a Call-ID: 1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250 CSeq: 104 BYE Via: SIP/2.0/UDP 161.49.142.250:5060;rport=5060;branch=z9hG4bK4ede3adc Supported: 100rel,sipvc,replaces Content-Length: 0 <-------------> hfemsrv*CLI> quit --- (8 headers 0 lines) --- hfemsrv*CLI> quit [Apr 19 14:26:09] WARNING[18442]: chan_sip.c:12311 handle_response: Remote host can't match request BYE to call '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250'. Giving up. hfemsrv*CLI> quit Really destroying SIP dialog '1a17a38a73f5418d5b23c2ab2c2623a8@161.49.142.250' Method: INVITE hfemsrv*CLI> quit Executing last minute cleanups Asterisk cleanly ending (0).