=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.03.25 19:56:32 =~=~=~=~=~=~=~=~=~=~=~= login as: root root@sip.commzilla.net's password: Last login: Tue Mar 25 19:47:24 2008 from 192.168.192.10 ]0;root@sip:~[root@sip ~]# asterisk -vvvvvvvvvr Asterisk 1.4.19, 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 Connected to Asterisk 1.4.19 currently running on sip (pid = 8202) sip*CLI> Verbosity is at least 18 Core debug is at least 10 sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: OPTIONS sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK43e8389a;rport From: "asterisk" ;tag=as6d0332b0 To: Contact: Call-ID: 68c9e09f0f5a42ba3109f84b1d9eb509@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:56:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK43e8389a;rport=5060;received=192.168.192.1 From: "asterisk" ;tag=as6d0332b0 To: ;tag=1789340700 Call-ID: 68c9e09f0f5a42ba3109f84b1d9eb509@192.168.192.1 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '68c9e09f0f5a42ba3109f84b1d9eb509@192.168.192.1' Method: OPTIONS sip*CLI> <--- SIP read from 66.49.255.61:5060 ---> OPTIONS sip:9057564513@99.242.44.13 SIP/2.0 Via: SIP/2.0/UDP 66.49.255.61:5060;branch=z9hG4bK60a76c1d;rport From: "asterisk" ;tag=as08d2652f To: Contact: Call-ID: 794d79e1367701d705d3f3207f918655@66.49.255.61 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:56:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Looking for 9057564513 in default (domain 99.242.44.13) <--- Transmitting (no NAT) to 66.49.255.61:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 66.49.255.61:5060;branch=z9hG4bK60a76c1d;received=66.49.255.61;rport=5060 From: "asterisk" ;tag=as08d2652f To: ;tag=as0f8d7691 Call-ID: 794d79e1367701d705d3f3207f918655@66.49.255.61 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '794d79e1367701d705d3f3207f918655@66.49.255.61' in 32000 ms (Method: OPTIONS) sip*CLI> Reliably Transmitting (no NAT) to 66.49.255.61:5060: OPTIONS sip:voip5-3.acanac.com SIP/2.0 Via: SIP/2.0/UDP 99.242.44.13:5060;branch=z9hG4bK00e9b1fa;rport From: "asterisk" ;tag=as556976af To: Contact: Call-ID: 7c7f33374d500c3425e2cc7579659550@99.242.44.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:56:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 66.49.255.61:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 99.242.44.13:5060;branch=z9hG4bK00e9b1fa;received=99.242.44.13;rport=5060 From: "asterisk" ;tag=as556976af To: ;tag=as1520df19 Call-ID: 7c7f33374d500c3425e2cc7579659550@99.242.44.13 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '7c7f33374d500c3425e2cc7579659550@99.242.44.13' Method: OPTIONS sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:220@sip.commzilla.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKbea5887569a3d5da4 Max-Forwards: 70 From: "Piano" ;tag=d5effe779b To: "220" Call-ID: 3781aa2bab331501 CSeq: 20226 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.2.0.166 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Using INVITE request as basis request - 3781aa2bab331501 <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKbea5887569a3d5da4;received=192.168.192.27 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as6f175b7c Call-ID: 3781aa2bab331501 CSeq: 20226 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="sip.commzilla.net", nonce="63e7aa7f" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3781aa2bab331501' in 32000 ms (Method: INVITE) Found user '220' sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:220@sip.commzilla.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bKbea5887569a3d5da4 Max-Forwards: 70 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as6f175b7c Call-ID: 3781aa2bab331501 CSeq: 20226 ACK User-Agent: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> INVITE sip:220@sip.commzilla.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6 Proxy-Authorization: Digest username="220",realm="sip.commzilla.net",nonce="63e7aa7f",uri="sip:220@sip.commzilla.net:5060",response="71031da8cc108be07b583532dc908840",algorithm=MD5 Max-Forwards: 70 From: "Piano" ;tag=d5effe779b To: "220" Call-ID: 3781aa2bab331501 CSeq: 20227 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Contact: Piano Session-Expires: 90 Supported: timer, 100rel, replaces User-Agent: Aastra 57i/2.2.0.166 Content-Type: application/sdp Content-Length: 599 v=0 o=MxSIP 0 0 IN IP4 192.168.192.27 s=SIP Call c=IN IP4 192.168.192.27 t=0 0 m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 25 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) Using INVITE request as basis request - 3781aa2bab331501 Found user '220' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 113 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 112 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 115 Found RTP audio format 96 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.192.27:3000 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found unknown media description format BV16 for ID 106 Found unknown media description format BV32 for ID 107 Found audio description format L16 for ID 113 Found audio description format PCMU for ID 110 Found audio description format PCMA for ID 111 Found audio description format L16 for ID 112 Found unknown media description format G726-16 for ID 98 Found unknown media description format G726-24 for ID 97 Found audio description format G726-32 for ID 115 Found unknown media description format G726-40 for ID 96 Found audio description format G722 for ID 9 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.27:3000 Looking for 220 in commzilla-super (domain sip.commzilla.net) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6;received=192.168.192.27 From: "Piano" ;tag=d5effe779b To: "220" Call-ID: 3781aa2bab331501 CSeq: 20227 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5a6e2526;rport From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Contact: Call-ID: a4a63b3e5f4f9a99 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- sip*CLI> Extension Changed 220[commzilla-local] new state InUse for Notify User 220 (queued) sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2163: NOTIFY sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7bef4cee;rport From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Contact: Call-ID: 3c26701547e6-d83kben73p4r CSeq: 187 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 confirmed --- sip*CLI> Extension Changed 220[commzilla-local] new state InUse for Notify User 233 sip*CLI> -- Executing [220@commzilla-super:1] GotoIf("SIP/220-096f99e0", "0?3") in new stack sip*CLI> -- Executing [220@commzilla-super:2] Set("SIP/220-096f99e0", "GROUP(OUTGOING)=220") in new stack sip*CLI> -- Executing [220@commzilla-super:3] Set("SIP/220-096f99e0", "OUTBOUND_GROUP_ONCE=220@INCOMING") in new stack sip*CLI> -- Executing [220@commzilla-super:4] Set("SIP/220-096f99e0", "DB(commzilla/wrapup/220/lastcall)=1206489420.114") in new stack sip*CLI> -- Executing [220@commzilla-super:5] Macro("SIP/220-096f99e0", "commzilla-dial|SIP/220|220|commzilla|20|en||twWkKM(all-tapi^1206489420.114)||default|||vm") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:1] NoOp("SIP/220-096f99e0", ""CALL TO LOCAL EXTENSION FROM 220(Piano)"") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:2] UserEvent("SIP/220-096f99e0", "TAPI|TAPIEVENT: LINE_NEWCALL commzilla") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:3] UserEvent("SIP/220-096f99e0", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:4] UserEvent("SIP/220-096f99e0", "TAPI|TAPIEVENT: SET CALLERID ") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:5] UserEvent("SIP/220-096f99e0", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:6] AGI("SIP/220-096f99e0", "/var/www/scopserv/telephony/scripts/agi/dial.php") in new stack sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK7bef4cee;rport=5060 From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Call-ID: 3c26701547e6-d83kben73p4r CSeq: 187 NOTIFY Content-Length: 0 <-------------> sip*CLI> --- (7 headers 0 lines) --- sip*CLI> SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/dial.php sip*CLI> -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5a6e2526;rport=5060;received=192.168.192.1 From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Call-ID: a4a63b3e5f4f9a99 CSeq: 106 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> sip*CLI> --- (9 headers 0 lines) --- sip*CLI> SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Extension State for '220' is '1'. sip*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) sip*CLI> -- /var/www/scopserv/telephony/scripts/agi/dial.php: Doing the action dial with params : 220 sip*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Dial string is SIP/220|20|twWkKM(all-tapi^1206489420.114)|. sip*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Possible Loop detected! Going to voicemail sip*CLI> -- /var/www/scopserv/telephony/scripts/agi/dial.php: Doing the action voicemail sip*CLI> -- /var/www/scopserv/telephony/scripts/agi/dial.php: Doing the action busy sip*CLI> -- AGI Script Executing Application: (Playtones) Options: (busy) sip*CLI> Audio is at 192.168.192.1 port 14192 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6;received=192.168.192.27 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as489e09af Call-ID: 3781aa2bab331501 CSeq: 20227 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 8202 8202 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 14192 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> sip*CLI> -- AGI Script /var/www/scopserv/telephony/scripts/agi/dial.php completed, returning 0 sip*CLI> -- Executing [220@commzilla-super:6] GotoIf("SIP/220-096f99e0", "0?7:8") in new stack sip*CLI> -- Goto (commzilla-super,220,8) sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> CANCEL sip:220@sip.commzilla.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6 Max-Forwards: 70 From: "Piano" ;tag=d5effe779b To: "220" Call-ID: 3781aa2bab331501 CSeq: 20227 CANCEL User-Agent: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.192.27 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6;received=192.168.192.27 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as489e09af Call-ID: 3781aa2bab331501 CSeq: 20227 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> sip*CLI> <--- Transmitting (no NAT) to 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6;received=192.168.192.27 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as489e09af Call-ID: 3781aa2bab331501 CSeq: 20227 CANCEL User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK17387e61;rport From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Contact: Call-ID: a4a63b3e5f4f9a99 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- sip*CLI> Extension Changed 220[commzilla-local] new state Idle for Notify User 220 (queued) sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2163: NOTIFY sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4dcd4f26;rport From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Contact: Call-ID: 3c26701547e6-d83kben73p4r CSeq: 188 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 209 terminated --- sip*CLI> Extension Changed 220[commzilla-local] new state Idle for Notify User 233 sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4dcd4f26;rport=5060 From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Call-ID: 3c26701547e6-d83kben73p4r CSeq: 188 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> ACK sip:220@sip.commzilla.net:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.192.27:5060;branch=z9hG4bK0193b82a09ac50ba6 Max-Forwards: 70 From: "Piano" ;tag=d5effe779b To: "220" ;tag=as489e09af Call-ID: 3781aa2bab331501 CSeq: 20227 ACK User-Agent: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '3781aa2bab331501' Method: ACK sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK17387e61;rport=5060;received=192.168.192.1 From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Call-ID: a4a63b3e5f4f9a99 CSeq: 107 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.28:5060: OPTIONS sip:225@192.168.192.28;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4ccbeae6;rport From: "asterisk" ;tag=as0e032577 To: Contact: Call-ID: 3d0bd6d7647a3f4c3f7684dc3cdfb518@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:57:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.28:5060 ---> SIP/2.0 200 OK Call-ID: 3d0bd6d7647a3f4c3f7684dc3cdfb518@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0e032577 To: ;tag=1acecc793dea318 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4ccbeae6;rport Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (11 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '3d0bd6d7647a3f4c3f7684dc3cdfb518@192.168.192.1' Method: OPTIONS sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@sip.commzilla.net SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK65261622;rport From: "Brillert" ;tag=1713567631 To: Supported: replaces, 100rel, timer Call-ID: 1876800092@192.168.192.24 CSeq: 20 INVITE Session-Expires: 1800 Contact: Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 273 v=0 o=222 1469577953 1183632275 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-11 <-------------> --- (14 headers 12 lines) --- Sending to 192.168.192.24 : 5060 (NAT) Using INVITE request as basis request - 1876800092@192.168.192.24 <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK65261622;received=192.168.192.24;rport=5060 From: "Brillert" ;tag=1713567631 To: ;tag=as0fa3f968 Call-ID: 1876800092@192.168.192.24 CSeq: 20 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="sip.commzilla.net", nonce="4d128657" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1876800092@192.168.192.24' in 32000 ms (Method: INVITE) Found user '222' sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@sip.commzilla.net SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK65261622;rport From: "Brillert" ;tag=1713567631 To: ;tag=as0fa3f968 Call-ID: 1876800092@192.168.192.24 CSeq: 20 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> INVITE sip:220@sip.commzilla.net SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563 From: "Brillert" ;tag=1713567631 To: Supported: replaces, 100rel, timer Call-ID: 1876800092@192.168.192.24 CSeq: 21 INVITE Session-Expires: 1800 Contact: Proxy-Authorization: Digest username="222", realm="sip.commzilla.net", nonce="4d128657", uri="sip:220@sip.commzilla.net", response="493e136e0f8348241504f5ba3a3c6388", algorithm=MD5 Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Type: application/sdp Content-Length: 273 v=0 o=222 1469577953 1183632275 IN IP4 192.168.192.24 s=A_converstion c=IN IP4 192.168.192.24 t=0 0 m=audio 15000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-11 <-------------> --- (15 headers 12 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) Using INVITE request as basis request - 1876800092@192.168.192.24 Found user '222' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Peer audio RTP is at port 192.168.192.24:15000 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 96 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.192.24:15000 Looking for 220 in commzilla-super (domain sip.commzilla.net) list_route: hop: <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563;received=192.168.192.24 From: "Brillert" ;tag=1713567631 To: Call-ID: 1876800092@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4bd20c47;rport From: ;tag=as04f438f1 To: Piano ;tag=c90f55300c Contact: Call-ID: 6c7f24fec1f43fee CSeq: 105 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- Extension Changed 222[commzilla-local] new state InUse for Notify User 220 sip*CLI> -- Executing [220@commzilla-super:1] GotoIf("SIP/222-096f99e0", "0?3") in new stack sip*CLI> -- Executing [220@commzilla-super:2] Set("SIP/222-096f99e0", "GROUP(OUTGOING)=222") in new stack sip*CLI> -- Executing [220@commzilla-super:3] Set("SIP/222-096f99e0", "OUTBOUND_GROUP_ONCE=220@INCOMING") in new stack sip*CLI> -- Executing [220@commzilla-super:4] Set("SIP/222-096f99e0", "DB(commzilla/wrapup/220/lastcall)=1206489432.115") in new stack sip*CLI> -- Executing [220@commzilla-super:5] Macro("SIP/222-096f99e0", "commzilla-dial|SIP/220|220|commzilla|20|en||twWkKM(all-tapi^1206489432.115)||default|||vm") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:1] NoOp("SIP/222-096f99e0", ""CALL TO LOCAL EXTENSION FROM 222(WiFi)"") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:2] UserEvent("SIP/222-096f99e0", "TAPI|TAPIEVENT: LINE_NEWCALL commzilla") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:3] UserEvent("SIP/222-096f99e0", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:4] UserEvent("SIP/222-096f99e0", "TAPI|TAPIEVENT: SET CALLERID ") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:5] UserEvent("SIP/222-096f99e0", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack sip*CLI> -- Executing [s@macro-commzilla-dial:6] AGI("SIP/222-096f99e0", "/var/www/scopserv/telephony/scripts/agi/dial.php") in new stack sip*CLI> -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/dial.php sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK4bd20c47;rport=5060;received=192.168.192.1 From: ;tag=as04f438f1 To: Piano ;tag=c90f55300c Call-ID: 6c7f24fec1f43fee CSeq: 105 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) sip*CLI> == Parsing '/etc/asterisk/manager.conf': Found sip*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Extension State for '220' is '0'. sip*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) sip*CLI> -- /var/www/scopserv/telephony/scripts/agi/dial.php: Doing the action dial with params : 220 sip*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Dial string is SIP/220|20|twWkKM(all-tapi^1206489432.115)|. sip*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/220|20|twWkKM(all-tapi^1206489432.115)|) sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK57697268;rport From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Contact: Call-ID: a4a63b3e5f4f9a99 CSeq: 108 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 230 early --- sip*CLI> Extension Changed 220[commzilla-local] new state Ringing for Notify User 220 (queued) sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2163: NOTIFY sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK13183002;rport From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Contact: Call-ID: 3c26701547e6-d83kben73p4r CSeq: 189 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 226 early --- sip*CLI> Extension Changed 220[commzilla-local] new state Ringing for Notify User 233 sip*CLI> Audio is at 192.168.192.1 port 14890 sip*CLI> Adding codec 0x4 (ulaw) to SDP sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.27:5060: INVITE sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport From: "WiFi" ;tag=as7fffa9ca To: Contact: Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 INVITE User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Date: Tue, 25 Mar 2008 23:57:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 8202 8202 IN IP4 192.168.192.1 s=session c=IN IP4 192.168.192.1 t=0 0 m=audio 14890 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sip*CLI> -- Called 220 sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK13183002;rport=5060 From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Call-ID: 3c26701547e6-d83kben73p4r CSeq: 189 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as7fffa9ca To: ;tag=4110370967 Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference Call-Info: ;appearance-index=1 Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- sip*CLI> -- SIP/220-096c4820 is ringing <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563;received=192.168.192.24 From: "Brillert" ;tag=1713567631 To: ;tag=as0a962547 Call-ID: 1876800092@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK57697268;rport=5060;received=192.168.192.1 From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Call-ID: a4a63b3e5f4f9a99 CSeq: 108 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> CANCEL sip:220@sip.commzilla.net SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563 From: "Brillert" ;tag=1713567631 To: Supported: replaces, 100rel, timer Call-ID: 1876800092@192.168.192.24 CSeq: 21 CANCEL Contact: Max-Forwards: 70 User-Agent: WirelessIP5000-v2.2.6/00:03:2a:00:53:68 Expires: 180 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.192.24 : 5060 (no NAT) <--- Reliably Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563;received=192.168.192.24 From: "Brillert" ;tag=1713567631 To: ;tag=as0a962547 Call-ID: 1876800092@192.168.192.24 CSeq: 21 INVITE User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.192.24:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563;received=192.168.192.24 From: "Brillert" ;tag=1713567631 To: ;tag=as0a962547 Call-ID: 1876800092@192.168.192.24 CSeq: 21 CANCEL User-Agent: Asterisk PBX (ScopServ) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> sip*CLI> Scheduling destruction of SIP dialog '57396ad853b89e0f2108aa4446eb76b9@192.168.192.1' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.192.27:5060: CANCEL sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport From: "WiFi" ;tag=as7fffa9ca To: Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 CANCEL User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- Scheduling destruction of SIP dialog '57396ad853b89e0f2108aa4446eb76b9@192.168.192.1' in 6400 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK77abfc1e;rport From: ;tag=as04f438f1 To: Piano ;tag=c90f55300c Contact: Call-ID: 6c7f24fec1f43fee CSeq: 106 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 222[commzilla-local] new state Idle for Notify User 220 Reliably Transmitting (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5f9e0897;rport From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Contact: Call-ID: a4a63b3e5f4f9a99 CSeq: 109 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- Extension Changed 220[commzilla-local] new state InUse for Notify User 220 (queued) Reliably Transmitting (no NAT) to 192.168.192.19:2163: NOTIFY sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5ebcf13b;rport From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Contact: Call-ID: 3c26701547e6-d83kben73p4r CSeq: 190 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 confirmed --- Extension Changed 220[commzilla-local] new state InUse for Notify User 233 sip*CLI> == Spawn extension (macro-commzilla-dial, s, 6) exited non-zero on 'SIP/222-096f99e0' -- Executing [h@macro-commzilla-dial:1] ResetCDR("SIP/222-096f99e0", "w") in new stack sip*CLI> -- Executing [h@macro-commzilla-dial:2] NoCDR("SIP/222-096f99e0", "") in new stack -- Executing [h@macro-commzilla-dial:3] UserEvent("SIP/222-096f99e0", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE") in new stack -- Executing [h@macro-commzilla-dial:4] System("SIP/222-096f99e0", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1206489432.115") in new stack sip*CLI> <--- SIP read from 192.168.192.24:5060 ---> ACK sip:220@sip.commzilla.net SIP/2.0 Via: SIP/2.0/UDP 192.168.192.24:5060;branch=z9hG4bK486584563 From: "Brillert" ;tag=1713567631 To: ;tag=as0a962547 Call-ID: 1876800092@192.168.192.24 CSeq: 21 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5ebcf13b;rport=5060 From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Call-ID: 3c26701547e6-d83kben73p4r CSeq: 190 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as7fffa9ca To: ;tag=4110370967 Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 CANCEL Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- sip*CLI> Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK77abfc1e;rport From: ;tag=as04f438f1 To: Piano ;tag=c90f55300c Contact: Call-ID: 6c7f24fec1f43fee CSeq: 106 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Retransmitting #1 (no NAT) to 192.168.192.27:5060: NOTIFY sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5f9e0897;rport From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Contact: Call-ID: a4a63b3e5f4f9a99 CSeq: 109 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport=5060;received=192.168.192.1 From: "WiFi" ;tag=as7fffa9ca To: ;tag=4110370967 Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 INVITE Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.192.27:5060: ACK sip:220@192.168.192.27:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK3dc7f3d2;rport From: "WiFi" ;tag=as7fffa9ca To: ;tag=4110370967 Contact: Call-ID: 57396ad853b89e0f2108aa4446eb76b9@192.168.192.1 CSeq: 102 ACK User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Remote-Party-ID: "WiFi" ;privacy=off;screen=no Content-Length: 0 --- sip*CLI> Extension Changed 220[commzilla-local] new state Idle for Notify User 220 (queued) Reliably Transmitting (no NAT) to 192.168.192.19:2163: NOTIFY sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK56031572;rport From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Contact: Call-ID: 3c26701547e6-d83kben73p4r CSeq: 191 NOTIFY User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 209 terminated --- Extension Changed 220[commzilla-local] new state Idle for Notify User 233 sip*CLI> Really destroying SIP dialog '57396ad853b89e0f2108aa4446eb76b9@192.168.192.1' Method: INVITE sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK56031572;rport=5060 From: ;tag=as024aad2f To: ;tag=f9lf9fspjh Call-ID: 3c26701547e6-d83kben73p4r CSeq: 191 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> Really destroying SIP dialog '1876800092@192.168.192.24' Method: ACK sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK77abfc1e;rport=5060;received=192.168.192.1 From: ;tag=as04f438f1 To: Piano ;tag=c90f55300c Call-ID: 6c7f24fec1f43fee CSeq: 106 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> <--- SIP read from 192.168.192.27:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5f9e0897;rport=5060;received=192.168.192.1 From: ;tag=as6d754962 To: Piano ;tag=5fe947b84b Call-ID: a4a63b3e5f4f9a99 CSeq: 109 NOTIFY Contact: Piano Server: Aastra 57i/2.2.0.166 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.19:2163: OPTIONS sip:233@192.168.192.19:2163;line=bk3dk6bw SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK22e07823;rport From: "asterisk" ;tag=as340c9e6d To: Contact: Call-ID: 4da806b3010bf00c49a9f9bb5b7d585d@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:57:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.19:2163 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK22e07823;rport=5060 From: "asterisk" ;tag=as340c9e6d To: Call-ID: 4da806b3010bf00c49a9f9bb5b7d585d@192.168.192.1 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/7.1.30 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Length: 0 <-------------> --- (14 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '4da806b3010bf00c49a9f9bb5b7d585d@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.26:5060: OPTIONS sip:223@192.168.192.26:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5862d602;rport From: "asterisk" ;tag=as174e5240 To: Contact: Call-ID: 064a22643c84527e700f0ad62d1ae1ae@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:57:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.26:1037 ---> SIP/2.0 405 Method Not Allowed Call-ID: 064a22643c84527e700f0ad62d1ae1ae@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as174e5240 To: ;tag=d30a38d4fb6e33d Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK5862d602;rport Content-Length: 0 Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REFER Allow: NOTIFY Allow: MESSAGE User-Agent: optiPoint 400 standard <-------------> --- (15 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '064a22643c84527e700f0ad62d1ae1ae@192.168.192.1' Method: OPTIONS sip*CLI> Reliably Transmitting (no NAT) to 192.168.192.21:5060: OPTIONS sip:226@192.168.192.21:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK655d2696;rport From: "asterisk" ;tag=as0deb5aee To: Contact: Call-ID: 639186d86b5c7622041a2f1d68f3a02c@192.168.192.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (ScopServ) Max-Forwards: 70 Date: Tue, 25 Mar 2008 23:57:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sip*CLI> <--- SIP read from 192.168.192.21:1129 ---> SIP/2.0 405 Method Not Allowed Call-ID: 639186d86b5c7622041a2f1d68f3a02c@192.168.192.1 CSeq: 102 OPTIONS From: "asterisk" ;tag=as0deb5aee To: ;tag=2f8bc292db63e6d Via: SIP/2.0/UDP 192.168.192.1:5060;branch=z9hG4bK655d2696;rport Content-Length: 0 Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REFER Allow: NOTIFY Allow: MESSAGE User-Agent: optiPoint 400 standard <-------------> --- (15 headers 0 lines) --- sip*CLI> Really destroying SIP dialog '639186d86b5c7622041a2f1d68f3a02c@192.168.192.1' Method: OPTIONS sip*CLI> ]0;root@sip:~[root@sip ~]# logout