call:/var/spool/asterisk# asterisk -r Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= Connected to Asterisk 1.2.13 currently running on call (pid = 16320) call*CLI> sip debug SIP Debugging enabled call*CLI> quit call:/var/spool/asterisk# ./makecall.sh call:/var/spool/asterisk# asterisk -r Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details. ========================================================================= Connected to Asterisk 1.2.13 currently running on call (pid = 16320) Feb 6 20:25:04 WARNING[16326]: pbx_spool.c:251 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/aiure1a.call: Operation not permitted We're at 80.86.113.155 port 18928 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: INVITE sip:0743049336@sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport From: "asterisk" ;tag=as487b799e To: Contact: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 102 INVITE User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 162 v=0 o=root 16320 16320 IN IP4 80.86.113.155 s=session c=IN IP4 80.86.113.155 t=0 0 m=audio 18928 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport=5060 From: "asterisk" ;tag=as487b799e To: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 102 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport=5060 Record-Route: From: asterisk ;tag=as487b799e To: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 102 INVITE Server: Sippy WWW-Authenticate: Digest realm="193.16.148.244",nonce="cf172c1f0030bf651f93f69cc8d8aa7347a9fb80" --- (9 headers 0 lines) --- Transmitting (no NAT) to 193.16.148.244:5060: ACK sip:0743049336@sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport From: "asterisk" ;tag=as487b799e To: Contact: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 102 ACK User-Agent: asterisk Max-Forwards: 70 Content-Length: 0 --- We're at 80.86.113.155 port 18928 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 193.16.148.244:5060: INVITE sip:0743049336@sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport From: "asterisk" ;tag=as487b799e To: Contact: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 103 INVITE User-Agent: asterisk Max-Forwards: 70 Authorization: Digest username="40215694650", realm="193.16.148.244", algorithm=MD5, uri="sip:0743049336@sip.netmaster.ro", nonce="cf172c1f0030bf651f93f69cc8d8aa7347a9fb80", response="0aac3635635f9e25cd38648e8486511d", opaque="" Date: Wed, 06 Feb 2008 18:25:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 162 v=0 o=root 16320 16321 IN IP4 80.86.113.155 s=session c=IN IP4 80.86.113.155 t=0 0 m=audio 18928 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060 From: "asterisk" ;tag=as487b799e To: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 103 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 --- (8 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK65842f90;rport From: "asterisk" ;tag=as77bba60c To: Contact: Call-ID: 24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK65842f90;rport=5060 From: asterisk ;tag=as77bba60c To: Call-ID: 24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155' call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060 Record-Route: From: asterisk ;tag=as487b799e To: ;tag=b88cae893f45b54799c6de2842e6596c Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 103 INVITE Server: Sippy --- (8 headers 0 lines) --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060 Record-Route: From: asterisk ;tag=as487b799e To: ;tag=b88cae893f45b54799c6de2842e6596c Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 103 INVITE Server: Sippy Contact: Anonymous Content-Length: 160 Content-Type: application/sdp =0ll*CLI> o=Sippy 139289996 1 IN IP4 193.16.148.244 s=SIP Call t=0 0 m=audio 19030 RTP/AVP 0 c=IN IP4 193.16.148.226 a=rtpmap:0 PCMU/8000 a=direction:passive --- (11 headers 8 lines) --- Found RTP audio format 0 Peer audio RTP is at port 193.16.148.226:19030 Found description format PCMU Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 193.16.148.244, port 5060 Transmitting (no NAT) to 193.16.148.244:5060: ACK sip:193.16.148.244:5061 SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK62119997;rport Route: From: "asterisk" ;tag=as487b799e To: ;tag=b88cae893f45b54799c6de2842e6596c Contact: Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 103 ACK User-Agent: asterisk Max-Forwards: 70 Content-Length: 0 --- Feb 6 20:25:13 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format Feb 6 20:25:13 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory Feb 6 20:25:13 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test Feb 6 20:25:14 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format Feb 6 20:25:14 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory Feb 6 20:25:14 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test !! The files are missing, this doesn't change the setup 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK07ed2d00;rport From: "asterisk" ;tag=as4f7286f4 To: Contact: Call-ID: 47793f687d077bf056b07cb20ffc5eae@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK07ed2d00;rport=5060 From: asterisk ;tag=as4f7286f4 To: Call-ID: 47793f687d077bf056b07cb20ffc5eae@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '47793f687d077bf056b07cb20ffc5eae@80.86.113.155' Feb 6 20:25:15 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format Feb 6 20:25:15 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory Feb 6 20:25:15 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test call*CLI> <-- SIP read from 193.16.148.244:5060: BYE sip:40215694650@80.86.113.155 SIP/2.0 Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKb087.ed99b927cbab958fd385005c501f5534.0 Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bKc4573e9d3872d522005ed7040940405d;rport=5061 Max-Forwards: 16 From: ;tag=b88cae893f45b54799c6de2842e6596c To: asterisk ;tag=as487b799e Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 100 BYE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 2400891340-3571454428-2228224023-2760872502 h323-conf-id: 2400891340-3571454428-2228224023-2760872502 !!! this is about where I hang up --- (13 headers 0 lines) --- Sending to 193.16.148.244 : 5060 (non-NAT) Transmitting (no NAT) to 193.16.148.244:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKb087.ed99b927cbab958fd385005c501f5534.0;received=193.16.148.244 Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bKc4573e9d3872d522005ed7040940405d;rport=5061 From: ;tag=b88cae893f45b54799c6de2842e6596c To: asterisk ;tag=as487b799e Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro CSeq: 100 BYE User-Agent: asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Feb 6 20:25:16 NOTICE[23141]: pbx_spool.c:279 attempt_thread: Call completed to SIP/0743049336@smartcall0 Destroying call '1a3884401ab152fd3d3e18fc3f855152@netmaster.ro' 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK1a5f44bd;rport From: "asterisk" ;tag=as0aed1590 To: Contact: Call-ID: 259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK1a5f44bd;rport=5060 From: asterisk ;tag=as0aed1590 To: Call-ID: 259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155' 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK309ae7a0;rport From: "asterisk" ;tag=as3f3a0b05 To: Contact: Call-ID: 6f93781a179da7dd7ab4d905272f5483@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK309ae7a0;rport=5060 From: asterisk ;tag=as3f3a0b05 To: Call-ID: 6f93781a179da7dd7ab4d905272f5483@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '6f93781a179da7dd7ab4d905272f5483@80.86.113.155' 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.136:5060: OPTIONS sip:*201@192.168.0.136:5060 SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5249fd38 From: "asterisk" ;tag=as641fa3f3 To: Contact: Call-ID: 388125593aac8ebf43a85b921454e2f4@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 192.168.0.136:5060: SIP/2.0 200 OK To: ;tag=81133be08fb82b0bi0 From: "asterisk" ;tag=as641fa3f3 Call-ID: 388125593aac8ebf43a85b921454e2f4@80.86.113.155 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5249fd38 Server: Linksys/SPA941-5.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces --- (10 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.0.136:5061: OPTIONS sip:*202@192.168.0.136:5061 SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK67951513 From: "asterisk" ;tag=as067cd32e To: Contact: Call-ID: 16db13571eee6f2d16aa15396506b26d@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '388125593aac8ebf43a85b921454e2f4@80.86.113.155' call*CLI> <-- SIP read from 192.168.0.136:5061: SIP/2.0 200 OK To: ;tag=e1756d38f5cfe993i1 From: "asterisk" ;tag=as067cd32e Call-ID: 16db13571eee6f2d16aa15396506b26d@80.86.113.155 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK67951513 Server: Linksys/SPA941-5.1.5 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces --- (10 headers 0 lines) --- Destroying call '16db13571eee6f2d16aa15396506b26d@80.86.113.155' 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5d3322f2;rport From: "asterisk" ;tag=as0fc21d1b To: Contact: Call-ID: 578bb0a4485bd3483a24f57909977c6e@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5d3322f2;rport=5060 From: asterisk ;tag=as0fc21d1b To: Call-ID: 578bb0a4485bd3483a24f57909977c6e@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '578bb0a4485bd3483a24f57909977c6e@80.86.113.155' 12 headers, 0 lines Reliably Transmitting (no NAT) to 193.16.148.244:5060: OPTIONS sip:sip.netmaster.ro SIP/2.0 Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK3d1085f9;rport From: "asterisk" ;tag=as5b319304 To: Contact: Call-ID: 5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155 CSeq: 102 OPTIONS User-Agent: asterisk Max-Forwards: 70 Date: Wed, 06 Feb 2008 18:25:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- call*CLI> <-- SIP read from 193.16.148.244:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK3d1085f9;rport=5060 From: asterisk ;tag=as5b319304 To: Call-ID: 5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155 CSeq: 102 OPTIONS Server: Sippy --- (7 headers 0 lines) --- Destroying call '5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155' call*CLI>