Password: root@1[root]# aa == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk CVS-HEAD-06/06/04-10:33:17, Copyright (C) 1999-2004 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD-06/06/04-10:33:17 currently running on asterisk (pid = 745) asterisk*CLI> sip debug SIP Debugging Enabled < snip > Sip read: INVITE sip:100@asterisk.travelinsured.net:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bka6375999cebc98 From:Joey+Stanford;tag=4C28AFB2993C171657 To: Call-ID:5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq:1 INVITE Supported: replaces,timer Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER Contact: Max-Forwards: 70 User-Agent: WiSIP Content-Type: application/sdp Content-Length: 205 v=0 o=TelogyUnknown0001 79991135 79991135 IN IP4 192.168.1.112 s=RTP Audio c=IN IP4 192.168.1.112 t=0 0 m=audio 2072 RTP/AVP 18 0 8 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 13 headers, 9 lines Using latest request as basis request Sending to 192.168.1.112 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Peer RTP is at port 192.168.1.112:0 Found description format G729 Found description format PCMU Found description format PCMA Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bka6375999cebc98;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as0101c77a Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7a45ccb8" Content-Length: 0 to 199.45.160.248:5060 Scheduling destruction of call '5542-D1B9-1281-F794-717A48BC7825@192.168.1.112' in 15000 ms Found user '101' asterisk*CLI> Sip read: ACK sip:100@asterisk.travelinsured.net:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bka6375999cebc98 From:Joey+Stanford;tag=4C28AFB2993C171657 To:;tag=as0101c77a Call-ID:5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq:1 ACK User-Agent: WiSIP Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines asterisk*CLI> Sip read: INVITE sip:100@asterisk.travelinsured.net:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149 From:Joey+Stanford;tag=4C28AFB2993C171657 To: Call-ID:5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq:2 INVITE Proxy-Authorization: DIGEST username="101",realm="asterisk",nonce="7a45ccb8",uri="sip:100@asterisk.travelinsured.net",algorithm=MD5,response="076245644cab563582e1e45b9123ad9a" Supported: replaces,timer Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,SUBSCRIBE,NOTIFY,INFO,REFER Contact: Max-Forwards: 70 User-Agent: WiSIP Content-Type: application/sdp Content-Length: 205 v=0 o=TelogyUnknown0001 79991135 79991135 IN IP4 192.168.1.112 s=RTP Audio c=IN IP4 192.168.1.112 t=0 0 m=audio 2072 RTP/AVP 18 0 8 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 14 headers, 9 lines Using latest request as basis request Sending to 192.168.1.112 : 5060 (NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Peer RTP is at port 192.168.1.112:0 Found description format G729 Found description format PCMU Found description format PCMA Capabilities: us - 0xe(GSM|ULAW|ALAW), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY) Found user '101' Looking for 100 in sip list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 199.45.160.248:5060 -- Executing Macro("SIP/101-c960", "stdexten|100|SIP/100&SIP/101") in new stack -- Executing Dial("SIP/101-c960", "SIP/100&SIP/101|20|TtrH") in new stack Destroying call '3f01c704477620c43cefdec72e5b0800@172.16.2.150' Jun 6 16:55:34 NOTICE[278542]: app_dial.c:674 dial_exec: Unable to create channel of type 'SIP' We're at 172.16.2.150 port 10618 Answering/Requesting with root capability 4 Answering with preferred capability 0x8(ALAW) Answering with preferred capability 0x2(GSM) Answering with non-codec capability 0x1(G723) 12 headers, 12 lines Reliably Transmitting: INVITE sip:101@199.45.160.248 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK71e86cf6 From: "Joey Stanford (WIFI)" ;tag=as6a6885d0 To: Contact: Call-ID: 6c83a1710d3e08e72d4de2131e0c6f0d@172.16.2.150 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:55:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 10618 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 199.45.160.248:5060 -- Called 101 Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 < snip > Sip read: SIP/2.0 100 Trying Via:SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK71e86cf6 From:Joey Stanford (WIFI);tag=as6a6885d0 To: Call-ID:6c83a1710d3e08e72d4de2131e0c6f0d@172.16.2.150 CSeq:102 INVITE Contact: User-Agent: WiSIP Content-Length: 0 9 headers, 0 lines asterisk*CLI> Sip read: SIP/2.0 486 Busy Via:SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK71e86cf6 From:Joey Stanford (WIFI);tag=as6a6885d0 To:;tag=BD1B68B15ECB61856F54 Call-ID:6c83a1710d3e08e72d4de2131e0c6f0d@172.16.2.150 CSeq:102 INVITE Contact: User-Agent: WiSIP Content-Length: 0 9 headers, 0 lines -- Got SIP response 486 "Busy" back from 199.45.160.248 Transmitting: ACK sip:192.168.1.112:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK71e86cf6 From: "Joey Stanford (WIFI)" ;tag=as6a6885d0 To: ;tag=BD1B68B15ECB61856F54 Contact: Call-ID: 6c83a1710d3e08e72d4de2131e0c6f0d@172.16.2.150 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 199.45.160.248:5060 -- SIP/101-f394 is busy == Everyone is busy at this time -- Executing VoiceMail("SIP/101-c960", "b100") in new stack We're at 172.16.2.150 port 19180 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Answering with preferred capability 0x2(GSM) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - to 199.45.160.248:5060 -- Playing 'voicemail/default/100/busy' (language 'en') Destroying call '6c83a1710d3e08e72d4de2131e0c6f0d@172.16.2.150' asterisk*CLI> < snip > asterisk*CLI> to 68.191.60.17:33721 Scheduling destruction of call 'f17921fb-e034858b@192.168.1.12' in 15000 ms Retransmitting #1 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - .2Q to 199.45.160.248:5060 asterisk*CLI> < snip > 8 headers, 0 lines Destroying call '63b57ee25ec830cb21c0a51622b745e7@172.16.2.150' Retransmitting #2 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - .2Q to 199.45.160.248:5060 asterisk*CLI> < snip ...other traffic between extensions that I'm cutting out > asterisk*CLI> to 68.191.60.17:33721 Scheduling destruction of call 'f17921fb-e034858b@192.168.1.12' in 15000 ms Retransmitting #3 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - .2Q to 199.45.160.248:5060 asterisk*CLI> < snip > to 68.191.60.17:33721 Scheduling destruction of call 'f17921fb-e034858b@192.168.1.12' in 15000 ms Retransmitting #4 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - .2Q to 199.45.160.248:5060 asterisk*CLI> < snip> to 68.209.241.68:5060 Scheduling destruction of call '4c0db2ff-666f83d8@192.168.1.5' in 15000 ms Retransmitting #5 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bkcbe8d580b45149;received=199.45.160.248 From: Joey+Stanford;tag=4C28AFB2993C171657 To: ;tag=as649bc6ff Call-ID: 5542-D1B9-1281-F794-717A48BC7825@192.168.1.112 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 205 v=0 o=root 2503 2503 IN IP4 172.16.2.150 s=session c=IN IP4 172.16.2.150 t=0 0 m=audio 19180 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - .2Q to 199.45.160.248:5060 11 headers, 2 lines < snip > to 68.209.241.68:5060 Scheduling destruction of call '4c0db2ff-666f83d8@192.168.1.5' in 15000 ms -- Playing 'vm-intro' (language 'en') asterisk*CLI> < snip > Sip read: 0 headers, 0 lines -- Playing 'beep' (language 'en') asterisk*CLI> < snip > 8 headers, 0 lines Destroying call '1ac957582a13395f096a36957774051a@172.16.2.150' -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav49, 0x819d1b8 -- x=1, open writing: /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: gsm, 0x819c818 -- x=2, open writing: /var/spool/asterisk/voicemail/default/100/INBOX/msg0004 format: wav, 0x81c8db8 asterisk*CLI> sip < snip > asterisk*CLI> sip no d (you can see typing here at the console...lots of registers and qualify action) to 68.191.60.17:33721 Scheduling destruction of call 'f17921fb-e034858b@192.168.1.12' in 15000 ms Jun 6 16:55:51 WARNING[278542]: app_voicemail.c:1377 play_and_record: No audio available on SIP/101-c960?? -- User hung up -- Executing Playback("SIP/101-c960", "something-terribly-wrong") in new stack -- Playing 'something-terribly-wrong' (language 'en') asterisk*CLI> sip no de < snip > to 68.209.241.68:5060 Scheduling destruction of call '4c0db2ff-666f83d8@192.168.1.5' in 15000 ms 11 headers, 0 lineso debug Reliably Transmitting: OPTIONS sip:192.246.69.223 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK63a575a0 From: "asterisk" ;tag=as4fcf5635 To: Contact: Call-ID: 6c357aa767f3ee8629bfa1f068ace33a@172.16.2.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:55:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.246.69.223:5060 -- Executing Playback("SIP/101-c960", "please-contact-tech-supt") in new stack -- Playing 'please-contact-tech-supt' (language 'en') asterisk*CLI> sip no debug < snip> (no NAT) to 68.209.241.68:5060 11 headers, 0 lineso debug Reliably Transmitting: OPTIONS sip:192.246.69.223 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK162a8c6a From: "asterisk" ;tag=as2d1264a9 To: Contact: Call-ID: 5a423fde04416c60247303a24677ba86@172.16.2.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:55:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.246.69.223:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.246.69.223 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK5e682129 From: "asterisk" ;tag=as2241bf94 To: Contact: Call-ID: 050f4caa0bd38ae22355d3707cf3da7a@172.16.2.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:55:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.246.69.223:5060 asterisk*CLI> sip no debug < snip > asterisk*CLI> sip no debug Sip read: SIP/2.0 404 non-invite response Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK162a8c6a From: "asterisk" ;tag=as2d1264a9 To: ;tag=ec5154884c15db89cef5049aa4f06328.cc29 Call-ID: 5a423fde04416c60247303a24677ba86@172.16.2.150 CSeq: 102 OPTIONS Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '5a423fde04416c60247303a24677ba86@172.16.2.150' asterisk*CLI> sip no debug Sip read: SIP/2.0 404 non-invite response Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK5e682129 From: "asterisk" ;tag=as2241bf94 To: ;tag=ec5154884c15db89cef5049aa4f06328.1493 Call-ID: 050f4caa0bd38ae22355d3707cf3da7a@172.16.2.150 CSeq: 102 OPTIONS Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '050f4caa0bd38ae22355d3707cf3da7a@172.16.2.150' Retransmitting #1 (no NAT): OPTIONS sip:192.246.69.223 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK63a575a0 From: "asterisk" ;tag=as4fcf5635 To: Contact: Call-ID: 6c357aa767f3ee8629bfa1f068ace33a@172.16.2.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:55:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 Ð to 192.246.69.223:5060 asterisk*CLI> sip no debug < snip > (no NAT) to 82.97.10.20:5060 -- Executing Hangup("SIP/101-c960", "") in new stack == Spawn extension (macro-stdexten, s, 105) exited non-zero on 'SIP/101-c960' in macro 'stdexten' == Spawn extension (sip, 100, 1) exited non-zero on 'SIP/101-c960' -- Executing Hangup("SIP/101-c960", "") in new stack == Spawn extension (sip, h, 1) exited non-zero on 'SIP/101-c960' asterisk*CLI> sip no debug to 68.209.241.68:5060 Scheduling destruction of call '4c0db2ff-666f83d8@192.168.1.5' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:101@199.45.160.248 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK402ec589 From: "asterisk" ;tag=as44391d77 To: Contact: Call-ID: 757cf00f54c6e1ad5c5ef7c27a8c3cb9@172.16.2.150 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 5/0 (NAT) to 199.45.160.248:5060 Scheduling destruction of call '757cf00f54c6e1ad5c5ef7c27a8c3cb9@172.16.2.150' in 15000 ms asterisk*CLI> sip no debug Sip read: SIP/2.0 400 Bad Request Via:SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK402ec589 From:asterisk;tag=as44391d77 To: Call-ID:757cf00f54c6e1ad5c5ef7c27a8c3cb9@172.16.2.150 CSeq:102 NOTIFY User-Agent: WiSIP 7 headers, 0 lines -- Got SIP response 400 "Bad Request" back from 199.45.160.248 Destroying call '757cf00f54c6e1ad5c5ef7c27a8c3cb9@172.16.2.150' asterisk*CLI> sip no debug < snip> to 68.209.241.68:5060 Scheduling destruction of call '4c0db2ff-666f83d8@192.168.1.5' in 15000 ms 11 headers, 0 lineso debug Reliably Transmitting: OPTIONS sip:192.246.69.223 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK612bb5cd From: "asterisk" ;tag=as4557215a To: Contact: Call-ID: 32c171ce6934987a763b5e5d3d7b3738@172.16.2.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sun, 06 Jun 2004 20:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 < snip> Sip read: SIP/2.0 404 non-invite response Via: SIP/2.0/UDP 172.16.2.150:5060;branch=z9hG4bK612bb5cd From: "asterisk" ;tag=as4557215a To: ;tag=ec5154884c15db89cef5049aa4f06328.6727 Call-ID: 32c171ce6934987a763b5e5d3d7b3738@172.16.2.150 CSeq: 102 OPTIONS Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 8 headers, 0 lines Destroying call '32c171ce6934987a763b5e5d3d7b3738@172.16.2.150' asterisk*CLI> sip no debug