Script started on Wed 19 Nov 2008 03:29:13 PM GMT [root@localhost admin]# asterisk -vvvvvvvvvvvvvvr Asterisk 1.4.18.1, 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.18.1 currently running on localhost (pid = 2671) localhost*CLI> Verbosity is at least 26 localhost*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'asteriskDial.exe' logged on from 192.168.16.184 Audio is at 192.168.16.4 port 12840 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.116:5060: INVITE sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport From: "steve" ;tag=as6847ee7c To: Contact: Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Nov 2008 15:29:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 2671 2671 IN IP4 192.168.16.4 s=session c=IN IP4 192.168.16.4 t=0 0 m=audio 12840 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport=5060;received=192.168.16.4 From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport=5060;received=192.168.16.4 From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Supported: timer, replaces Content-Type: application/sdp Content-Length: 261 v=0 o=MxSIP 0 0 IN IP4 192.168.16.116 s=SIP Call c=IN IP4 192.168.16.116 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.16.116:3000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.16.116:3000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.116, port 5060 Transmitting (no NAT) to 192.168.16.116:5060: ACK sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK428be6e1;rport From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Contact: Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> > Channel SIP/220-08238318 was answered. == Manager 'asteriskDial.exe' logged off from 192.168.16.184 localhost*CLI> -- Executing [222@tapi:1] Macro("SIP/220-08238318", "stdexten|222|SIP/222") in new stack localhost*CLI> -- Executing [s@macro-stdexten:1] Dial("SIP/220-08238318", "SIP/222|20") in new stack Audio is at 192.168.16.4 port 16082 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.162:5060: INVITE sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: Contact: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Nov 2008 15:29:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 2671 2671 IN IP4 192.168.16.4 s=session c=IN IP4 192.168.16.4 t=0 0 m=audio 16082 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 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 222 localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Jonathan Holloway Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- SIP/222-082398a8 is ringing localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> BYE sip:asterisk@192.168.16.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK5157d57792324ae71.a24084db6060750a6 Max-Forwards: 70 From: ;tag=3421637935 To: "steve" ;tag=as6847ee7c Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 23318 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Supported: timer User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.16.116 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK5157d57792324ae71.a24084db6060750a6;received=192.168.16.116 From: ;tag=3421637935 To: "steve" ;tag=as6847ee7c Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 23318 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.16.162:5060: CANCEL sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' in 32000 ms (Method: INVITE) == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/220-08238318' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/220-08238318' localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 CANCEL Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.16.162:5060: ACK sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Contact: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Really destroying SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' Method: INVITE Really destroying SIP dialog '3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4' Method: BYE localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> REGISTER sip:192.168.16.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525 Max-Forwards: 70 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="222",realm="asterisk",nonce="2136bd24",uri="sip:192.168.16.4:5060",response="b3a4d263211b0fb2ed54ed2fcf831f2b",algorithm=MD5 Contact: "Jonathan Holloway" User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> localhost*CLI> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.16.162 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525;received=192.168.16.162 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525;received=192.168.16.162 From: ;tag=b1302b481c To: ;tag=as7f858b1e Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="214f9c85" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'd9a52942714a16a3' in 32000 ms (Method: REGISTER) localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> REGISTER sip:192.168.16.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28 Max-Forwards: 70 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="222",realm="asterisk",nonce="214f9c85",uri="sip:192.168.16.4:5060",response="c08dddb537d708d481924ca15b4dfec4",algorithm=MD5 Contact: "Jonathan Holloway" User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.16.162 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28;received=192.168.16.162 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28;received=192.168.16.162 From: ;tag=b1302b481c To: ;tag=as7f858b1e Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Wed, 19 Nov 2008 15:29:32 GMT Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog 'd9a52942714a16a3' in 32000 ms (Method: REGISTER) localhost*CLI> Scheduling destruction of SIP dialog '4475e34b72b96fac5b91dc472b73b63e@192.168.16.4' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 192.168.16.162:5060: NOTIFY sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK7ee8f473;rport From: "asterisk" ;tag=as677eeb89 To: Contact: Call-ID: 4475e34b72b96fac5b91dc472b73b63e@192.168.16.4 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@192.168.16.4 Voice-Message: 0/9 (0/0) --- localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK7ee8f473;rport=5060;received=192.168.16.4 From: "asterisk" ;tag=as677eeb89 To: ;tag=2610638338 Call-ID: 4475e34b72b96fac5b91dc472b73b63e@192.168.16.4 CSeq: 102 NOTIFY Contact: Jonathan Holloway Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- localhost*CLI> Really destroying SIP dialog '4475e34b72b96fac5b91dc472b73b63e@192.168.16.4' Method: NOTIFY localhost*CLI> quti localhost*CLI> No such command 'quti' (type 'help' for help) localhost*CLI> quit Executing last minute cleanups [root@localhost admin]# exit Script done on Wed 19 Nov 2008 03:29:42 PM GMT t admin]# [root@localhost admin]# [root@localhost admin]# [root@localhost admin]# [root@localhost admin]# ls typescript [root@localhost admin]# ls=dir typescript [root@localhost admin]# ls typescript [root@localhost admin]# script Script started, file is typescript [root@localhost admin]# asterisk -vvvvvvvvvvvvvvr Asterisk 1.4.18.1, 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. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.18.1 currently running on localhost (pid = 2671) localhost*CLI> Verbosity is at least 26 localhost*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'asteriskDial.exe' logged on from 192.168.16.184 Audio is at 192.168.16.4 port 12840 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.116:5060: INVITE sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport From: "steve" ;tag=as6847ee7c To: Contact: Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Nov 2008 15:29:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 2671 2671 IN IP4 192.168.16.4 s=session c=IN IP4 192.168.16.4 t=0 0 m=audio 12840 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport=5060;received=192.168.16.4 From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK641a8e99;rport=5060;received=192.168.16.4 From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Supported: timer, replaces Content-Type: application/sdp Content-Length: 261 v=0 o=MxSIP 0 0 IN IP4 192.168.16.116 s=SIP Call c=IN IP4 192.168.16.116 t=0 0 m=audio 3000 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.16.116:3000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.16.116:3000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.116, port 5060 Transmitting (no NAT) to 192.168.16.116:5060: ACK sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK428be6e1;rport From: "steve" ;tag=as6847ee7c To: ;tag=3421637935 Contact: Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> > Channel SIP/220-08238318 was answered. == Manager 'asteriskDial.exe' logged off from 192.168.16.184 localhost*CLI> -- Executing [222@tapi:1] Macro("SIP/220-08238318", "stdexten|222|SIP/222") in new stack localhost*CLI> -- Executing [s@macro-stdexten:1] Dial("SIP/220-08238318", "SIP/222|20") in new stack Audio is at 192.168.16.4 port 16082 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.162:5060: INVITE sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: Contact: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Nov 2008 15:29:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 2671 2671 IN IP4 192.168.16.4 s=session c=IN IP4 192.168.16.4 t=0 0 m=audio 16082 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 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 222 localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Jonathan Holloway Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- SIP/222-082398a8 is ringing localhost*CLI> <--- SIP read from 192.168.16.116:5060 ---> BYE sip:asterisk@192.168.16.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK5157d57792324ae71.a24084db6060750a6 Max-Forwards: 70 From: ;tag=3421637935 To: "steve" ;tag=as6847ee7c Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 23318 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Supported: timer User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.16.116 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK5157d57792324ae71.a24084db6060750a6;received=192.168.16.116 From: ;tag=3421637935 To: "steve" ;tag=as6847ee7c Call-ID: 3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4 CSeq: 23318 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.16.162:5060: CANCEL sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' in 32000 ms (Method: INVITE) == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/220-08238318' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/220-08238318' localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 CANCEL Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport=5060;received=192.168.16.4 From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 INVITE Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.16.162:5060: ACK sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK5758d807;rport From: "steve" ;tag=as15a7a8c1 To: ;tag=1337409839 Contact: Call-ID: 26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> Really destroying SIP dialog '26206c047e30503e61cfe6ae1a5b2c84@192.168.16.4' Method: INVITE Really destroying SIP dialog '3ad90a6b25e9b807646353317a6fe6a5@192.168.16.4' Method: BYE localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> REGISTER sip:192.168.16.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525 Max-Forwards: 70 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="222",realm="asterisk",nonce="2136bd24",uri="sip:192.168.16.4:5060",response="b3a4d263211b0fb2ed54ed2fcf831f2b",algorithm=MD5 Contact: "Jonathan Holloway" User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> localhost*CLI> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.16.162 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525;received=192.168.16.162 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bK6a1b013a2817560e7.6035fb3cdcb077525;received=192.168.16.162 From: ;tag=b1302b481c To: ;tag=as7f858b1e Call-ID: d9a52942714a16a3 CSeq: 25995 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="214f9c85" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'd9a52942714a16a3' in 32000 ms (Method: REGISTER) localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> REGISTER sip:192.168.16.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28 Max-Forwards: 70 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="222",realm="asterisk",nonce="214f9c85",uri="sip:192.168.16.4:5060",response="c08dddb537d708d481924ca15b4dfec4",algorithm=MD5 Contact: "Jonathan Holloway" User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.16.162 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28;received=192.168.16.162 From: ;tag=b1302b481c To: Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- Transmitting (no NAT) to 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.162:5060;branch=z9hG4bKb74e6fcd8a532dbe5.cf0b1b0eef1ab7f28;received=192.168.16.162 From: ;tag=b1302b481c To: ;tag=as7f858b1e Call-ID: d9a52942714a16a3 CSeq: 25996 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Wed, 19 Nov 2008 15:29:32 GMT Content-Length: 0 <------------> localhost*CLI> Scheduling destruction of SIP dialog 'd9a52942714a16a3' in 32000 ms (Method: REGISTER) localhost*CLI> Scheduling destruction of SIP dialog '4475e34b72b96fac5b91dc472b73b63e@192.168.16.4' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 192.168.16.162:5060: NOTIFY sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK7ee8f473;rport From: "asterisk" ;tag=as677eeb89 To: Contact: Call-ID: 4475e34b72b96fac5b91dc472b73b63e@192.168.16.4 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@192.168.16.4 Voice-Message: 0/9 (0/0) --- localhost*CLI> <--- SIP read from 192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.4:5060;branch=z9hG4bK7ee8f473;rport=5060;received=192.168.16.4 From: "asterisk" ;tag=as677eeb89 To: ;tag=2610638338 Call-ID: 4475e34b72b96fac5b91dc472b73b63e@192.168.16.4 CSeq: 102 NOTIFY Contact: Jonathan Holloway Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- localhost*CLI> Really destroying SIP dialog '4475e34b72b96fac5b91dc472b73b63e@192.168.16.4' Method: NOTIFY Script done on Wed 19 Nov 2008 03:32:58 PM GMT