=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2008.01.14 11:08:55 =~=~=~=~=~=~=~=~=~=~=~= root root@10.0.0.138's password: Last login: Mon Jan 14 10:55:11 2008 from 192.168.30.190 ]0;root@lab:~[root@lab ~]# asterisk -vvvvvvvvvvvvvvvvvvvvr Asterisk 1.4.17, Copyright (C) 1999 - 2007 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 1.4.17 currently running on lab (pid = 8834) lab*CLI> Verbosity is at least 20 lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK19704fbd;rport From: "asterisk" ;tag=as56ad5634 To: Contact: Call-ID: 40388bda190414a95fd407143df8b5f1@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #1 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK19704fbd;rport From: "asterisk" ;tag=as56ad5634 To: Contact: Call-ID: 40388bda190414a95fd407143df8b5f1@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #2 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK19704fbd;rport From: "asterisk" ;tag=as56ad5634 To: Contact: Call-ID: 40388bda190414a95fd407143df8b5f1@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Really destroying SIP dialog '3c267015ced7-oe17m9n64ttc' Method: REGISTER lab*CLI> Retransmitting #3 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK19704fbd;rport From: "asterisk" ;tag=as56ad5634 To: Contact: Call-ID: 40388bda190414a95fd407143df8b5f1@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #4 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK19704fbd;rport From: "asterisk" ;tag=as56ad5634 To: Contact: Call-ID: 40388bda190414a95fd407143df8b5f1@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Really destroying SIP dialog '40388bda190414a95fd407143df8b5f1@192.168.30.254' Method: OPTIONS lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INVITE sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-w0f3o2u167hf;rport From: "Reception 6000" ;tag=0lvocvkac8 To: Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 421 v=0 o=root 222634346 222634346 IN IP4 192.168.30.162 s=call c=IN IP4 192.168.30.162 t=0 0 m=audio 47988 RTP/AVP 0 8 9 2 3 18 4 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:E67orp+MMi2z8rTfOsDbDyFkeVeFClSVHoXg+CZ5 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 17 lines) --- [Jan 14 11:09:21] WARNING[8924]: rtp.c:1987 ast_rtp_settos: Unable to set TOS to 184 Sending to 192.168.30.162 : 2087 (NAT) Using INVITE request as basis request - 3c267040d7f1-251pjg8vpth8 <--- Reliably Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-w0f3o2u167hf;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as414dd6a0 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 1 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="fireworx", nonce="0916b6b2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c267040d7f1-251pjg8vpth8' in 32000 ms (Method: INVITE) Found user '6000' lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> ACK sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-w0f3o2u167hf;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as414dd6a0 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INVITE sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ls0i049ag6pw;rport From: "Reception 6000" ;tag=0lvocvkac8 To: Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="6000",realm="fireworx",nonce="0916b6b2",uri="sip:2221@192.168.30.254",response="2e928ee7934d498ac8bfcfe1c762c217",algorithm=MD5 Content-Type: application/sdp Content-Length: 421 v=0 o=root 222634346 222634346 IN IP4 192.168.30.162 s=call c=IN IP4 192.168.30.162 lab*CLI> t=0 0 m=audio 47988 RTP/AVP 0 8 9 2 3 18 4 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:E67orp+MMi2z8rTfOsDbDyFkeVeFClSVHoXg+CZ5 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 17 lines) --- Sending to 192.168.30.162 : 2087 (NAT) Using INVITE request as basis request - 3c267040d7f1-251pjg8vpth8 Found user '6000' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Peer audio RTP is at port 192.168.30.162:47988 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 2 Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) lab*CLI> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.30.162:47988 [Jan 14 11:09:21] DEBUG[8924]: chan_sip.c:3250 update_call_counter: Call from peer '6000' is 1 out of 4 Looking for 2221 in default-super (domain 192.168.30.254) lab*CLI> list_route: hop: lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ls0i049ag6pw;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK74a21acd;rport From: ;tag=as15e3080b To: ;tag=5vs24umnws Contact: Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 104 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206 confirmed --- lab*CLI> Extension Changed 6000 new state InUse for Notify User 6000 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK487e9e0c;rport From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Contact: Call-ID: 97f0e33a014c2888 CSeq: 140 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- lab*CLI> Extension Changed 6000 new state InUse for Notify User 6001 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK025bbf30;rport From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Contact: Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 141 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 confirmed --- lab*CLI> Extension Changed 6000 new state InUse for Notify User 6003 Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e435c3e;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 Contact: Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 CSeq: 145 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- Extension Changed 6000 new state InUse for Notify User 6002 lab*CLI> -- Executing [2221@default-super:1] Macro("SIP/6000-b7c52430", "default-dial-zap-5946cf-000148") in new stack lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:1] Set("SIP/6000-b7c52430", "DYNAMIC_FEATURES=automon#hookflash") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:2] Set("SIP/6000-b7c52430", "TOUCH_MONITOR=1200326961.108") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [s@macro-default-dial-zap-5946cf-000148:3] Set("SIP/6000-b7c52430", "DB(default/6000/RepeatDial)=2221") in new stack lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7e435c3e;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 CSeq: 145 NOTIFY Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:4] Set("SIP/6000-b7c52430", "DB(default/wrapup/s/lastcall)=1200326961.108") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [s@macro-default-dial-zap-5946cf-000148:5] Set("SIP/6000-b7c52430", "org_cidnum=6000") in new stack [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:6] Set("SIP/6000-b7c52430", "GROUP(OUTGOING)=6000") in new stack [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK025bbf30;rport=5060 From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 141 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Executing [s@macro-default-dial-zap-5946cf-000148:7] GotoIf("SIP/6000-b7c52430", "1?10") in new stack -- Goto (macro-default-dial-zap-5946cf-000148,s,10) [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf -- Executing [s@macro-default-dial-zap-5946cf-000148:10] GotoIf("SIP/6000-b7c52430", "1?11:20") in new stack -- Goto (macro-default-dial-zap-5946cf-000148,s,11) [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf -- Executing [s@macro-default-dial-zap-5946cf-000148:11] Set("SIP/6000-b7c52430", "ext_cidname=God Calling") in new stack [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [s@macro-default-dial-zap-5946cf-000148:12] Set("SIP/6000-b7c52430", "ext_cidnum=6000") in new stack [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: func_db.c:70 function_db_read: DB: default/6000/CallerExternalPres not found in database. -- Executing [s@macro-default-dial-zap-5946cf-000148:13] Set("SIP/6000-b7c52430", "ext_cidpres=") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:14] GotoIf("SIP/6000-b7c52430", "0?16") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:15] Set("SIP/6000-b7c52430", "CALLERID(number)=6000") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:16] GotoIf("SIP/6000-b7c52430", "0?18") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:17] Set("SIP/6000-b7c52430", "CALLERID(name)=God Calling") in new stack lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Set lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:18] GotoIf("SIP/6000-b7c52430", "1?20") in new stack lab*CLI> -- Goto (macro-default-dial-zap-5946cf-000148,s,20) lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:20] Goto("SIP/6000-b7c52430", "50") in new stack lab*CLI> -- Goto (macro-default-dial-zap-5946cf-000148,s,50) lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: Goto lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:50] GotoIf("SIP/6000-b7c52430", "1?52") in new stack lab*CLI> -- Goto (macro-default-dial-zap-5946cf-000148,s,52) lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:52] GotoIf("SIP/6000-b7c52430", "0?53:55") in new stack lab*CLI> -- Goto (macro-default-dial-zap-5946cf-000148,s,55) lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:55] GotoIf("SIP/6000-b7c52430", "0?56:57") in new stack lab*CLI> -- Goto (macro-default-dial-zap-5946cf-000148,s,57) lab*CLI> [Jan 14 11:09:21] DEBUG[10877]: app_macro.c:337 _macro_exec: Executed application: GotoIf lab*CLI> -- Executing [s@macro-default-dial-zap-5946cf-000148:57] Dial("SIP/6000-b7c52430", "Zap/g1/2221|60|TtwW") in new stack lab*CLI> -- Requested transfer capability: 0x00 - SPEECH lab*CLI> -- Called g1/2221 lab*CLI> Audio is at 192.168.30.254 port 19462 lab*CLI> Adding codec 0x4 (ulaw) to SDP lab*CLI> Adding codec 0x100 (g729) to SDP lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ls0i049ag6pw;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 8834 8834 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19462 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK487e9e0c;rport=5060;received=192.168.30.254 From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Call-ID: 97f0e33a014c2888 CSeq: 140 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK74a21acd;rport=5060 From: ;tag=as15e3080b To: ;tag=5vs24umnws Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 104 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> [Jan 14 11:09:22] DEBUG[9129]: chan_zap.c:9020 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/1 span 1 -- Zap/1-1 is proceeding passing it to SIP/6000-b7c52430 lab*CLI> -- Zap/1-1 is ringing <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ls0i049ag6pw;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:22] DEBUG[9129]: chan_zap.c:1420 zt_enable_ec: Echo cancellation already on lab*CLI> -- Zap/1-1 answered SIP/6000-b7c52430 Audio is at 192.168.30.254 port 19462 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP <--- Reliably Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ls0i049ag6pw;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 8834 8835 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 19462 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> ACK sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-60418gx9v0hh;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> lab*CLI> --- (9 headers 0 lines) --- lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b527d1e;rport From: "asterisk" ;tag=as65d68b72 To: Contact: Call-ID: 0914d8953cf473322e2f56734d94f0b6@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-e3nl4n79z7sk;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 3 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-e3nl4n79z7sk;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 3 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-blubr37y4vzw;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 4 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-blubr37y4vzw;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 4 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:24] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 160 queued on SIP/6000-b7c52430 [Jan 14 11:09:24] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:24] DTMF[10877]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-cgpr179fug1y;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 5 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-cgpr179fug1y;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 5 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 14 11:09:24] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:24] DTMF[10877]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ns7nzj7u4lpg;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 6 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 lab*CLI> <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ns7nzj7u4lpg;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 6 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:24] DTMF[10877]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2254 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on Zap/1-1, duration 0 ms [Jan 14 11:09:24] DTMF[10877]: channel.c:2427 __ast_read: DTMF end accepted without begin '9' on Zap/1-1 [Jan 14 11:09:24] DTMF[10877]: channel.c:2438 __ast_read: DTMF end passthrough '9' on Zap/1-1 lab*CLI> [Jan 14 11:09:24] DEBUG[10877]: chan_sip.c:5871 reqprep: Strict routing enforced for session 3c267040d7f1-251pjg8vpth8 lab*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.30.162, port 2087 Reliably Transmitting (no NAT) to 192.168.30.162:2087: INFO sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7b9620b0;rport From: ;tag=as06c3c682 To: "Reception 6000" ;tag=0lvocvkac8 Contact: Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 102 INFO User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 24 Signal=9 Duration=100 --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7b9620b0;rport=5060 From: ;tag=as06c3c682 To: "Reception 6000" ;tag=0lvocvkac8 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 102 INFO Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog INFO arrived lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2254 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:24] DTMF[10877]: channel.c:2254 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/6000-b7c52430 lab*CLI> Retransmitting #1 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b527d1e;rport From: "asterisk" ;tag=as65d68b72 To: Contact: Call-ID: 0914d8953cf473322e2f56734d94f0b6@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> [Jan 14 11:09:25] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:25] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '*' lab*CLI> [Jan 14 11:09:25] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '*' lab*CLI> [Jan 14 11:09:25] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '9' lab*CLI> [Jan 14 11:09:25] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '9' lab*CLI> [Jan 14 11:09:26] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '9' lab*CLI> Retransmitting #2 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b527d1e;rport From: "asterisk" ;tag=as65d68b72 To: Contact: Call-ID: 0914d8953cf473322e2f56734d94f0b6@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> [Jan 14 11:09:26] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '9' lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> Retransmitting #3 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b527d1e;rport From: "asterisk" ;tag=as65d68b72 To: Contact: Call-ID: 0914d8953cf473322e2f56734d94f0b6@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #4 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6b527d1e;rport From: "asterisk" ;tag=as65d68b72 To: Contact: Call-ID: 0914d8953cf473322e2f56734d94f0b6@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Really destroying SIP dialog '0914d8953cf473322e2f56734d94f0b6@192.168.30.254' Method: OPTIONS lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ogecbx97xizy;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 7 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ogecbx97xizy;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 7 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:28] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:28] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 160 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:28] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-pse38z1gbt3u;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 8 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-pse38z1gbt3u;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 8 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '9' with duration 160 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '*' lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-wu81zxh7dfk8;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 9 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-wu81zxh7dfk8;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 9 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:29] DTMF[9896]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:29] DTMF[9896]: channel.c:2438 __ast_read: DTMF end passthrough '9' on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '*' lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '9' lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-wdgtvfc57asz;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 10 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 lab*CLI> <-------------> lab*CLI> --- (11 headers 2 lines) --- Receiving INFO! lab*CLI> * DTMF-relay event received: 9 lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-wdgtvfc57asz;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 10 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '9' lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2254 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '9' lab*CLI> [Jan 14 11:09:29] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '9' lab*CLI> [Jan 14 11:09:30] DTMF[10877]: channel.c:2254 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:30] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:30] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '9' lab*CLI> [Jan 14 11:09:30] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '9' lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-tuz472y3g6of;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 11 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 lab*CLI> <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-tuz472y3g6of;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 11 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:35] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:35] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 160 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:35] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:35] DEBUG[10877]: chan_zap.c:1055 zt_digit_begin: Started VLDTMF digit '*' lab*CLI> [Jan 14 11:09:35] DEBUG[10877]: chan_zap.c:1090 zt_digit_end: Ending VLDTMF digit '*' lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-3p70ob8k8oy3;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 12 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 lab*CLI> <-------------> lab*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-3p70ob8k8oy3;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 12 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:36] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:36] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 160 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:36] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-naseklydzx1q;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 13 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=2 Duration=160 lab*CLI> <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 2 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-naseklydzx1q;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 13 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:36] DTMF[10877]: channel.c:2381 __ast_read: DTMF end '2' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:36] DTMF[10877]: channel.c:2417 __ast_read: DTMF begin emulation of '2' with duration 160 queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:37] DTMF[10877]: channel.c:2501 __ast_read: DTMF end emulation of '2' queued on SIP/6000-b7c52430 -- Started music on hold, class 'default', on channel 'Zap/1-1' lab*CLI> -- Playing 'pbx-transfer' (language 'en') lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK709ed03f;rport From: "asterisk" ;tag=as4fd0329c To: Contact: Call-ID: 34e0d3070b4c960d2116c5492407f2b8@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #1 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK709ed03f;rport From: "asterisk" ;tag=as4fd0329c To: Contact: Call-ID: 34e0d3070b4c960d2116c5492407f2b8@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #2 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK709ed03f;rport From: "asterisk" ;tag=as4fd0329c To: Contact: Call-ID: 34e0d3070b4c960d2116c5492407f2b8@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> BYE sip:2221@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-kzd0520sr7c0;rport From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 14 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 RTP-RxStat: Total_Rx_Pkts=812,Rx_Pkts=812,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=893,Tx_Pkts=893,Remote_Tx_Pkts=662 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.30.162 : 2087 (NAT) <--- Transmitting (NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-kzd0520sr7c0;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=0lvocvkac8 To: ;tag=as06c3c682 Call-ID: 3c267040d7f1-251pjg8vpth8 CSeq: 14 BYE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> -- Stopped music on hold on Zap/1-1 lab*CLI> [Jan 14 11:09:40] DEBUG[10877]: chan_zap.c:2972 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 lab*CLI> [Jan 14 11:09:40] DEBUG[10877]: chan_zap.c:2611 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call lab*CLI> [Jan 14 11:09:40] DEBUG[10877]: chan_zap.c:2968 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 lab*CLI> -- Hungup 'Zap/1-1' == Spawn extension (macro-default-dial-zap-5946cf-000148, s, 57) exited non-zero on 'SIP/6000-b7c52430' in macro 'default-dial-zap-5946cf-000148' == Spawn extension (macro-default-dial-zap-5946cf-000148, s, 57) exited non-zero on 'SIP/6000-b7c52430' -- Executing [h@macro-default-dial-zap-5946cf-000148:1] ResetCDR("SIP/6000-b7c52430", "w") in new stack -- Executing [h@macro-default-dial-zap-5946cf-000148:2] NoCDR("SIP/6000-b7c52430", "") in new stack -- Executing [h@macro-default-dial-zap-5946cf-000148:3] System("SIP/6000-b7c52430", "/var/www/scopserv/telephony/scripts/billing/cdr.sh 1200326961.108") in new stack lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.162:2087: OPTIONS sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4bdc5695;rport From: "asterisk" ;tag=as5cacdbd6 To: Contact: Call-ID: 74499edd25d99745081755db7ac05bde@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK4bdc5695;rport=5060 From: "asterisk" ;tag=as5cacdbd6 To: Call-ID: 74499edd25d99745081755db7ac05bde@192.168.30.254 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom360/7.1.30 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) --- lab*CLI> Really destroying SIP dialog '74499edd25d99745081755db7ac05bde@192.168.30.254' Method: OPTIONS lab*CLI> [Jan 14 11:09:40] DEBUG[10877]: chan_sip.c:3224 update_call_counter: Call from peer '6000' removed from call limit 4 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK656d987d;rport From: ;tag=as15e3080b To: ;tag=5vs24umnws Contact: Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 105 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- lab*CLI> Extension Changed 6000 new state Idle for Notify User 6000 Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK200246e8;rport From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Contact: Call-ID: 97f0e33a014c2888 CSeq: 141 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- lab*CLI> Extension Changed 6000 new state Idle for Notify User 6001 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK77251abe;rport From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Contact: Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 142 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 terminated --- lab*CLI> Extension Changed 6000 new state Idle for Notify User 6003 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0adccc64;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 Contact: Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 CSeq: 146 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 348
--- lab*CLI> Extension Changed 6000 new state Idle for Notify User 6002 lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK0adccc64;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 CSeq: 146 NOTIFY Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK77251abe;rport=5060 From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 142 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK656d987d;rport=5060 From: ;tag=as15e3080b To: ;tag=5vs24umnws Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 105 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> Really destroying SIP dialog '3c267040d7f1-251pjg8vpth8' Method: BYE lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK200246e8;rport=5060;received=192.168.30.254 From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Call-ID: 97f0e33a014c2888 CSeq: 141 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> Retransmitting #3 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK709ed03f;rport From: "asterisk" ;tag=as4fd0329c To: Contact: Call-ID: 34e0d3070b4c960d2116c5492407f2b8@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #4 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK709ed03f;rport From: "asterisk" ;tag=as4fd0329c To: Contact: Call-ID: 34e0d3070b4c960d2116c5492407f2b8@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '34e0d3070b4c960d2116c5492407f2b8@192.168.30.254' Method: OPTIONS lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INVITE sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-i9gxlf1bb7f7;rport From: "Reception 6000" ;tag=v2ps0x737t To: Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 421 v=0 o=root 198994046 198994046 IN IP4 192.168.30.162 s=call c=IN IP4 192.168.30.162 t=0 0 m=audio 43294 RTP/AVP 0 8 9 2 3 18 4 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JDY1+eteGx+O+94GNwzDKz6UqkPMIsnXk6BkDIZj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (18 headers 17 lines) --- [Jan 14 11:09:46] WARNING[8924]: rtp.c:1987 ast_rtp_settos: Unable to set TOS to 184 Sending to 192.168.30.162 : 2087 (NAT) Using INVITE request as basis request - 3c267059a6ad-x0wqz0vxxeey <--- Reliably Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-i9gxlf1bb7f7;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as7eae40ca Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 1 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="fireworx", nonce="32f501ef" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c267059a6ad-x0wqz0vxxeey' in 32000 ms (Method: INVITE) Found user '6000' lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> ACK sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-i9gxlf1bb7f7;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as7eae40ca Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INVITE sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-sncljxbug6ow;rport From: "Reception 6000" ;tag=v2ps0x737t To: Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/7.1.30 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 Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="6000",realm="fireworx",nonce="32f501ef",uri="sip:6002@192.168.30.254",response="99bebf1859b60c93a9cb5dd85238cfb6",algorithm=MD5 Content-Type: application/sdp Content-Length: 421 v=0 o=root 198994046 198994046 IN IP4 192.168.30.162 s=call c=IN IP4 192.168.30.162 lab*CLI> t=0 0 m=audio 43294 RTP/AVP 0 8 9 2 3 18 4 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JDY1+eteGx+O+94GNwzDKz6UqkPMIsnXk6BkDIZj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=ptime:20 a=encryption:optional a=sendrecv <-------------> --- (19 headers 17 lines) --- Sending to 192.168.30.162 : 2087 (NAT) Using INVITE request as basis request - 3c267059a6ad-x0wqz0vxxeey Found user '6000' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Peer audio RTP is at port 192.168.30.162:43294 Found audio description format pcmu for ID 0 Found audio description format pcma for ID 8 Found audio description format g722 for ID 9 Found audio description format g726-32 for ID 2 lab*CLI> Found audio description format gsm for ID 3 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.30.162:43294 [Jan 14 11:09:46] DEBUG[8924]: chan_sip.c:3250 update_call_counter: Call from peer '6000' is 1 out of 4 Looking for 6002 in default-super (domain 192.168.30.254) Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK127cf910;rport From: ;tag=as15e3080b To: ;tag=5vs24umnws Contact: Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206 confirmed --- lab*CLI> list_route: hop: Extension Changed 6000 new state InUse for Notify User 6000 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-sncljxbug6ow;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK40c087a8;rport From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Contact: Call-ID: 97f0e33a014c2888 CSeq: 142 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 212 confirmed --- lab*CLI> Extension Changed 6000 new state InUse for Notify User 6001 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2e47ba07;rport From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Contact: Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 143 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 confirmed --- lab*CLI> Extension Changed 6000 new state InUse for Notify User 6003 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK11065fdf;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 Contact: Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 CSeq: 147 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- Extension Changed 6000 new state InUse for Notify User 6002 -- Executing [6002@default-super:1] GotoIf("SIP/6000-b7c52430", "0?3") in new stack lab*CLI> -- Executing [6002@default-super:2] Set("SIP/6000-b7c52430", "GROUP(OUTGOING)=6000") in new stack lab*CLI> -- Executing [6002@default-super:3] Set("SIP/6000-b7c52430", "OUTBOUND_GROUP_ONCE=6002@INCOMING") in new stack lab*CLI> -- Executing [6002@default-super:4] Set("SIP/6000-b7c52430", "DB(default/wrapup/6002/lastcall)=1200326986.110") in new stack lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK11065fdf;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 CSeq: 147 NOTIFY Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> lab*CLI> --- (10 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK2e47ba07;rport=5060 From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 143 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> -- Executing [6002@default-super:5] Macro("SIP/6000-b7c52430", "default-dial|SIP/6002|6002|default|20|en|6002@default|twWkKM(all-tapi^1200326986.110)||default||Local/0@default-local|Local/s@default-aa-servicemainmenubutton1englishsubmenu") in new stack lab*CLI> -- Executing [s@macro-default-dial:1] NoOp("SIP/6000-b7c52430", ""CALL TO LOCAL EXTENSION FROM 6000(Reception 6000)"") in new stack lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: NoOp lab*CLI> -- Executing [s@macro-default-dial:2] UserEvent("SIP/6000-b7c52430", "TAPI|TAPIEVENT: LINE_NEWCALL default") in new stack lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: UserEvent lab*CLI> -- Executing [s@macro-default-dial:3] UserEvent("SIP/6000-b7c52430", "TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_OFFERING") in new stack lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: UserEvent lab*CLI> -- Executing [s@macro-default-dial:4] UserEvent("SIP/6000-b7c52430", "TAPI|TAPIEVENT: SET CALLERID ") in new stack lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: UserEvent lab*CLI> -- Executing [s@macro-default-dial:5] UserEvent("SIP/6000-b7c52430", "TAPI|TAPIEVENT: LINE_CALLINFO LINECALLINFOSTATE_CALLERID") in new stack lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: UserEvent lab*CLI> -- Executing [s@macro-default-dial:6] AGI("SIP/6000-b7c52430", "/var/www/scopserv/telephony/scripts/agi/dial.php") in new stack lab*CLI> -- Launched AGI Script /var/www/scopserv/telephony/scripts/agi/dial.php lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK40c087a8;rport=5060;received=192.168.30.254 From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Call-ID: 97f0e33a014c2888 CSeq: 142 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> lab*CLI> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK127cf910;rport=5060 From: ;tag=as15e3080b To: ;tag=5vs24umnws Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 106 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Extension State for '6002' is '0'. lab*CLI> -- AGI Script Executing Application: (NoOp) Options: (STATUS:) lab*CLI> -- /var/www/scopserv/telephony/scripts/agi/dial.php: Doing the action dial with params : 6002 lab*CLI> /var/www/scopserv/telephony/scripts/agi/dial.php: Dial string is SIP/6002|20|twWkKM(all-tapi^1200326986.110)|. lab*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/6002|20|twWkKM(all-tapi^1200326986.110)|) lab*CLI> [Jan 14 11:09:46] WARNING[10921]: rtp.c:1987 ast_rtp_settos: Unable to set TOS to 184 lab*CLI> [Jan 14 11:09:46] DEBUG[10921]: chan_sip.c:3250 update_call_counter: Call to peer '6002' is 1 out of 4 Audio is at 192.168.30.254 port 16734 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.30.200:5060: INVITE sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6c4534cd;rport From: "Reception 6000" ;tag=as1284cac9 To: Contact: Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 CSeq: 102 INVITE User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Remote-Party-ID: "Reception 6000" ;privacy=off;screen=no Date: Mon, 14 Jan 2008 16:09:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 242 v=0 o=root 8834 8834 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 16734 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 --- -- Called 6002 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72daf722;rport From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Contact: Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 104 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 224 early --- Extension Changed 6002 new state Ringing for Notify User 6000 Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK300a9fff;rport From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 Contact: Call-ID: efff1cb032e8806b CSeq: 110 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 229 early --- lab*CLI> Extension Changed 6002 new state Ringing for Notify User 6001 Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK38c477c2;rport From: ;tag=as1880928a To: ;tag=1nr9f24wfd Contact: Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 116 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog C lab*CLI> ontent-Type: application/dialog-info+xml Subscription-State: active Content-Length: 225 early --- Extension Changed 6002 new state Ringing for Notify User 6003 Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5272028b;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 Contact: Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 CSeq: 133 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- Extension Changed 6002 new state Ringing for Notify User 6002 lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6c4534cd;rport From: "Reception 6000" ;tag=as1284cac9 To: ;tag=2955F5BE-D8D4A0E7 CSeq: 102 INVITE Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> lab*CLI> --- (9 headers 0 lines) --- lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK38c477c2;rport=5060 From: ;tag=as1880928a To: ;tag=1nr9f24wfd Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 116 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72daf722;rport=5060 From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 104 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5272028b;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 CSeq: 133 NOTIFY Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK300a9fff;rport=5060;received=192.168.30.254 From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 Call-ID: efff1cb032e8806b CSeq: 110 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6c4534cd;rport From: "Reception 6000" ;tag=as1284cac9 To: ;tag=2955F5BE-D8D4A0E7 CSeq: 102 INVITE Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- lab*CLI> -- SIP/6002-0918b7a0 is ringing <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-sncljxbug6ow;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK6c4534cd;rport From: "Reception 6000" ;tag=as1284cac9 To: ;tag=2955F5BE-D8D4A0E7 CSeq: 102 INVITE Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Type: application/sdp Content-Length: 203 v=0 o=- 1200326984 1200326984 IN IP4 192.168.30.200 s=Polycom IP Phone c=IN IP4 192.168.30.200 t=0 0 m=audio 2248 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.30.200:2248 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/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 192.168.30.200:2248 lab*CLI> list_route: hop: [Jan 14 11:09:47] DEBUG[8924]: chan_sip.c:5871 reqprep: Strict routing enforced for session 404d1d9d4961f632065669643485bfd7@192.168.30.254 set_destination: Parsing for address/port to send to Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7b171abb;rport From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Contact: Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 105 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206 confirmed --- Extension Changed 6002 new state InUse for Notify User 6000 set_destination: set destination to 192.168.30.200, port 5060 Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1aee2d1c;rport From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 lab*CLI> Contact: Call-ID: efff1cb032e8806b CSeq: 111 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 confirmed --- Extension Changed 6002 new state InUse for Notify User 6001 Transmitting (no NAT) to 192.168.30.200:5060: ACK sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK12c46341;rport From: "Reception 6000" ;tag=as1284cac9 To: ;tag=2955F5BE-D8D4A0E7 Contact: Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 CSeq: 102 ACK User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Remote-Party-ID: "Reception 6000" ;privacy=off;screen=no Content-Length: 0 --- lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7628d259;rport From: ;tag=as1880928a To: ;tag=1nr9f24wfd Contact: Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 117 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 confirmed --- -- SIP/6002-0918b7a0 answered SIP/6000-b7c52430 Extension Changed 6002 new state InUse for Notify User 6003 Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK010bbd9a;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 Contact: Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 CSeq: 134 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 353
--- -- Executing [s@macro-all-tapi:1] UserEvent("SIP/6002-0918b7a0", "TAPI|TAPIEVENT [~1200326986.110] LINE_CALLSTATE LINECALLSTATE_CONNECTED") in new stack Extension Changed 6002 new state InUse for Notify User 6002 lab*CLI> [Jan 14 11:09:47] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: UserEvent [Jan 14 11:09:47] DEBUG[10921]: app_dial.c:1575 dial_exec_full: Macro exited with status 0 lab*CLI> Audio is at 192.168.30.254 port 18210 Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP <--- Reliably Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-sncljxbug6ow;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 2 INVITE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 233 v=0 o=root 8834 8834 IN IP4 192.168.30.254 s=session c=IN IP4 192.168.30.254 t=0 0 m=audio 18210 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK010bbd9a;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 CSeq: 134 NOTIFY Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7628d259;rport=5060 From: ;tag=as1880928a To: ;tag=1nr9f24wfd Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 117 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7b171abb;rport=5060 From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 105 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1aee2d1c;rport=5060;received=192.168.30.254 From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 Call-ID: efff1cb032e8806b CSeq: 111 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> lab*CLI> --- (8 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> ACK sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-m0jpgp8pp5vo;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-q6zx5qryx33h;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 3 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 23 Signal=* Duration=9280 lab*CLI> <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-q6zx5qryx33h;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 3 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-kev043xq4004;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 4 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 23 Signal=9 Duration=9280 lab*CLI> <-------------> lab*CLI> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-kev043xq4004;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 4 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:49] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 9280 ms lab*CLI> [Jan 14 11:09:49] DTMF[10921]: channel.c:2417 __ast_read: DTMF begin emulation of '*' with duration 9280 queued on SIP/6000-b7c52430 [Jan 14 11:09:49] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 9280 ms [Jan 14 11:09:49] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-98dyxs6ozp54;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 5 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 23 Signal=9 Duration=9280 lab*CLI> <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-98dyxs6ozp54;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 5 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 14 11:09:49] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 9280 ms [Jan 14 11:09:49] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-gwmxkpy44u58;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 6 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 23 Signal=9 Duration=9280 <-------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-gwmxkpy44u58;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 6 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 14 11:09:49] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 9280 ms [Jan 14 11:09:49] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7dbb6fcc;rport From: "asterisk" ;tag=as625f3b03 To: Contact: Call-ID: 5572f1537b4c88980363de747e243a52@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #1 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7dbb6fcc;rport From: "asterisk" ;tag=as625f3b03 To: Contact: Call-ID: 5572f1537b4c88980363de747e243a52@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-hmjl6j90rnio;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 7 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 < lab*CLI> -------------> lab*CLI> --- (11 headers 2 lines) --- lab*CLI> Receiving INFO! lab*CLI> * DTMF-relay event received: * lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-hmjl6j90rnio;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 7 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:53] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '*' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:53] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '*' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-fpwye2ney9c2;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 8 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: 9 <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-fpwye2ney9c2;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 8 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:53] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms lab*CLI> [Jan 14 11:09:53] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ja7nowf62yb8;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 9 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 < lab*CLI> -------------> lab*CLI> --- (11 headers 2 lines) --- lab*CLI> Receiving INFO! lab*CLI> * DTMF-relay event received: 9 lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-ja7nowf62yb8;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 9 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:53] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:53] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> INFO sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-1l8csrl70hqe;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 10 INFO Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 Content-Type: application/dtmf-relay Content-Length: 22 Signal=9 Duration=160 lab*CLI> <-------------> lab*CLI> --- (11 headers 2 lines) --- lab*CLI> Receiving INFO! lab*CLI> * DTMF-relay event received: 9 lab*CLI> <--- Transmitting (no NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-1l8csrl70hqe;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 10 INFO User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:53] DTMF[10921]: channel.c:2381 __ast_read: DTMF end '9' received on SIP/6000-b7c52430, duration 160 ms [Jan 14 11:09:53] DTMF[10921]: channel.c:2387 __ast_read: DTMF end '9' put into dtmf queue on SIP/6000-b7c52430 lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> Retransmitting #2 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7dbb6fcc;rport From: "asterisk" ;tag=as625f3b03 To: Contact: Call-ID: 5572f1537b4c88980363de747e243a52@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #3 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7dbb6fcc;rport From: "asterisk" ;tag=as625f3b03 To: Contact: Call-ID: 5572f1537b4c88980363de747e243a52@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Retransmitting #4 (no NAT) to 192.168.30.136:5060: OPTIONS sip:192.168.30.136 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK7dbb6fcc;rport From: "asterisk" ;tag=as625f3b03 To: Contact: Call-ID: 5572f1537b4c88980363de747e243a52@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:09:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> Really destroying SIP dialog '5572f1537b4c88980363de747e243a52@192.168.30.254' Method: OPTIONS lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> BYE sip:6000@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.200:5060;branch=z9hG4bK4eb4ba8c4B1E403D From: ;tag=2955F5BE-D8D4A0E7 To: "Reception 6000" ;tag=as1284cac9 CSeq: 1 BYE Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Max-Forwards: 70 Content-Length: 0 <-------------> lab*CLI> --- (10 headers 0 lines) --- Sending to 192.168.30.200 : 5060 (no NAT) <--- Transmitting (no NAT) to 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.200:5060;branch=z9hG4bK4eb4ba8c4B1E403D;received=192.168.30.200 From: ;tag=2955F5BE-D8D4A0E7 To: "Reception 6000" ;tag=as1284cac9 Call-ID: 404d1d9d4961f632065669643485bfd7@192.168.30.254 CSeq: 1 BYE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> [Jan 14 11:09:57] DEBUG[10921]: chan_sip.c:3224 update_call_counter: Call to peer '6002' removed from call limit 4 lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72cfa92b;rport From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Contact: Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 106 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- Extension Changed 6002 new state Idle for Notify User 6000 Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK68bba168;rport From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 Contact: Call-ID: efff1cb032e8806b CSeq: 112 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 6002 new state Idle for Notify User 6001 Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK11fce05b;rport From: ;tag=as1880928a To: ;tag=1nr9f24wfd Contact: Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 118 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 terminated --- Extension Changed 6002 new state Idle for Notify User 6003 Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 V lab*CLI> ia: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5d787fc7;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 Contact: Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 CSeq: 135 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 348
--- Extension Changed 6002 new state Idle for Notify User 6002 -- AGI Script /var/www/scopserv/telephony/scripts/agi/dial.php completed, returning 0 [Jan 14 11:09:57] DEBUG[10921]: app_macro.c:337 _macro_exec: Executed application: AGI lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5d787fc7;rport From: ;tag=as065e173c To: "Reception 6002" ;tag=755D981B-644EF370 CSeq: 135 NOTIFY Call-ID: c2a8e904-a95053d5-d121382@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> lab*CLI> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> Really destroying SIP dialog '404d1d9d4961f632065669643485bfd7@192.168.30.254' Method: BYE lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK11fce05b;rport=5060 From: ;tag=as1880928a To: ;tag=1nr9f24wfd Call-ID: 3c2670203e82-ge91iif19r3o CSeq: 118 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK72cfa92b;rport=5060 From: ;tag=as0c4a5cc4 To: ;tag=2hlx6mxxqg Call-ID: 3c2670160e49-5c4mo7f2rd06 CSeq: 106 NOTIFY Content-Length: 0 <-------------> lab*CLI> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK68bba168;rport=5060;received=192.168.30.254 From: ;tag=as19da82ce To: Reception 6001 ;tag=ffe3b4dcd7 Call-ID: efff1cb032e8806b CSeq: 112 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2501 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> -- Invalid extension '*99' in context 'default-super' on SIP/6000-b7c52430 lab*CLI> == CDR updated on SIP/6000-b7c52430 lab*CLI> -- Executing [i@default-super:1] Playback("SIP/6000-b7c52430", "invalid") in new stack lab*CLI> -- Playing 'invalid' (language 'en') lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '*' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> [Jan 14 11:09:58] DTMF[10921]: channel.c:2251 __ast_read: DTMF end emulation of '9' queued on SIP/6000-b7c52430 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> BYE sip:6002@192.168.30.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-3dtm6popzvw8;rport From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 11 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom360/7.1.30 RTP-RxStat: Total_Rx_Pkts=491,Rx_Pkts=491,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=548,Tx_Pkts=548,Remote_Tx_Pkts=489 Content-Length: 0 <-------------> lab*CLI> --- (12 headers 0 lines) --- lab*CLI> Sending to 192.168.30.162 : 2087 (NAT) lab*CLI> <--- Transmitting (NAT) to 192.168.30.162:2087 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.162:2087;branch=z9hG4bK-3dtm6popzvw8;received=192.168.30.162;rport=2087 From: "Reception 6000" ;tag=v2ps0x737t To: ;tag=as57ac0a8d Call-ID: 3c267059a6ad-x0wqz0vxxeey CSeq: 11 BYE User-Agent: Asterisk PBX (Fireworx) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> lab*CLI> == Spawn extension (default-super, i, 1) exited non-zero on 'SIP/6000-b7c52430' [Jan 14 11:09:58] DEBUG[10921]: chan_sip.c:3224 update_call_counter: Call from peer '6000' removed from call limit 4 Reliably Transmitting (no NAT) to 192.168.30.162:2087: NOTIFY sip:6000@192.168.30.162:2087;line=5452o401 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK662908ad;rport From: ;tag=as15e3080b To: ;tag=5vs24umnws Contact: Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 107 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- lab*CLI> Extension Changed 6000 new state Idle for Notify User 6000 Reliably Transmitting (no NAT) to 192.168.30.193:5060: NOTIFY sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK33c4b52f;rport From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Contact: Call-ID: 97f0e33a014c2888 CSeq: 143 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 213 terminated --- Extension Changed 6000 new state Idle for Notify User 6001 Reliably Transmitting (no NAT) to 192.168.30.199:1046: NOTIFY sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1fad452c;rport From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Contact: Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 144 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 208 terminated --- Extension Changed 6000 new state Idle for Notify User 6003 Reliably Transmitting (no NAT) to 192.168.30.200:5060: NOTIFY sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK18d3bfa7;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 Contact: Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 CSeq: 148 NOTIFY User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 348
--- Extension Changed 6000 new state Idle for Notify User 6002 lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK18d3bfa7;rport From: ;tag=as61cb867a To: "Reception 6002" ;tag=F570E265-895599D2 CSeq: 148 NOTIFY Call-ID: 64cdadc6-2aa2a9cf-4372cd4@192.168.30.200 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> lab*CLI> --- (10 headers 0 lines) --- lab*CLI> SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> Really destroying SIP dialog '3c267059a6ad-x0wqz0vxxeey' Method: BYE lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK1fad452c;rport=5060 From: ;tag=as314e8d8e To: ;tag=rg47iqt9w3 Call-ID: 3c2670203b58-j6r2fngncv40 CSeq: 144 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 500 CSeq Number Out of order Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK33c4b52f;rport=5060;received=192.168.30.254 From: ;tag=as1bd164c1 To: Reception 6001 ;tag=75fba40d63 Call-ID: 97f0e33a014c2888 CSeq: 143 NOTIFY Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived -- Incoming call: Got SIP response 500 "CSeq Number Out of order" back from 192.168.30.193 lab*CLI> <--- SIP read from 192.168.30.162:2087 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK662908ad;rport=5060 From: ;tag=as15e3080b To: ;tag=5vs24umnws Call-ID: 3c2670160b16-t0yefuq9mw53 CSeq: 107 NOTIFY Content-Length: 0 <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.198:5060: OPTIONS sip:6007@192.168.30.198;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5fdf64d4;rport From: "asterisk" ;tag=as374c672e To: Contact: Call-ID: 45128cd92568404a4f8e4013469fa181@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:10:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.198:5060 ---> SIP/2.0 200 OK Call-ID: 45128cd92568404a4f8e4013469fa181@192.168.30.254 CSeq: 102 OPTIONS From: "asterisk" ;tag=as374c672e To: ;tag=80ed46c0ddf7d1e Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK5fdf64d4;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) --- lab*CLI> Really destroying SIP dialog '45128cd92568404a4f8e4013469fa181@192.168.30.254' Method: OPTIONS lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.199:1046: OPTIONS sip:6003@192.168.30.199:1046;line=guwdxbyt SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ea7c20b;rport From: "asterisk" ;tag=as58710e08 To: Contact: Call-ID: 1b3241dc73d4ca2b3c1f16580903ef08@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:10:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.199:1046 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK3ea7c20b;rport=5060 From: "asterisk" ;tag=as58710e08 To: Call-ID: 1b3241dc73d4ca2b3c1f16580903ef08@192.168.30.254 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom370/7.1.30 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) --- lab*CLI> Really destroying SIP dialog '1b3241dc73d4ca2b3c1f16580903ef08@192.168.30.254' Method: OPTIONS lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.200:5060: OPTIONS sip:6002@192.168.30.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK67b034a2;rport From: "asterisk" ;tag=as4c71a856 To: Contact: Call-ID: 34d7ddc5695d9d9c6b784e296e259949@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:10:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.200:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK67b034a2;rport From: "asterisk" ;tag=as4c71a856 To: ;tag=ADD499C3-791A1378 CSeq: 102 OPTIONS Call-ID: 34d7ddc5695d9d9c6b784e296e259949@192.168.30.254 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.1.0062 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- lab*CLI> Really destroying SIP dialog '34d7ddc5695d9d9c6b784e296e259949@192.168.30.254' Method: OPTIONS lab*CLI> Reliably Transmitting (no NAT) to 192.168.30.193:5060: OPTIONS sip:6001@192.168.30.193:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK198f993e;rport From: "asterisk" ;tag=as726dfda0 To: Contact: Call-ID: 357d50693f4b3e036a2ff0e6391c9681@192.168.30.254 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (Fireworx) Max-Forwards: 70 Date: Mon, 14 Jan 2008 16:10:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- lab*CLI> <--- SIP read from 192.168.30.193:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.30.254:5060;branch=z9hG4bK198f993e;rport=5060;received=192.168.30.254 From: "asterisk" ;tag=as726dfda0 To: ;tag=1651216401 Call-ID: 357d50693f4b3e036a2ff0e6391c9681@192.168.30.254 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/2.1.2.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- lab*CLI> Really destroying SIP dialog '357d50693f4b3e036a2ff0e6391c9681@192.168.30.254' Method: OPTIONS lab*CLI> == Parsing '/etc/asterisk/manager.conf': Found lab*CLI> ]0;root@lab:~[root@lab ~]# logout