Really destroying SIP dialog '1243735216@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> INVITE sip:00393337974334@87.247.224.11:5060 SIP/2.0 Record-Route: Max-Forwards: 68 Session-Expires: 3600;Refresher=uac Supported: timer, 100rel To: From: "1234" ;tag=3416295864-555413 Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Contact: Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=1000" Content-Type: application/sdp Content-Length: 226 X-Enswitch-Source: 213.166.9.34:5060 X-Enswitch-External: yes v=0 o=MSX2 2147483647 1 IN IP4 213.166.9.34 s=sip call c=IN IP4 213.166.9.26 t=0 0 m=audio 25764 RTP/AVP 18 101 a=ptime:30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (18 headers 11 lines) --- Sending to 87.247.224.5 : 5060 (NAT) Using INVITE request as basis request - 2644-3416295864-555405@MSX2.mydomain.com Found peer 'enswitch-local' Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 213.166.9.26:25764 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 213.166.9.26:25764 Looking for 00393337974334 in from-internal (domain 87.247.224.11) list_route: hop: <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Record-Route: From: "1234" ;tag=3416295864-555413 To: Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [00393337974334@from-internal:1] AGI("SIP/5060-088fb6e0", "agi://127.0.0.1/enswitch?stype=internal") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: enswitch?stype=internal AGI Tx >> agi_request: agi://127.0.0.1/enswitch?stype=internal AGI Tx >> agi_channel: SIP/5060-088fb6e0 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1207307011.895 AGI Tx >> agi_callerid: 1234 AGI Tx >> agi_calleridname: 1234 AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 00393337974334 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from-internal AGI Tx >> agi_extension: 00393337974334 AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> CLI> AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE __ENSWITCH-CALLID "1207307011.895" AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CUTOFF AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-SCREEN AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-DIVERTED AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE TIMEOUT(absolute) 86400 -- Channel will hangup at 2008-04-05 11:03:31 UTC. AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-SCUSTOMER AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(X-Enswitch-Source) AGI Tx >> 200 result=1 (213.166.9.34:5060) AGI Rx << GET VARIABLE ENSWITCH-PRESENTATION AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(X-Enswitch-External) AGI Tx >> 200 result=1 (yes) AGI Rx << GET VARIABLE ENSWITCH-TRANSFER-PHONE AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-TRANSFER-CUSTOMER AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Resolving type '', number '00393337974334'." 5 agi://127.0.0.1/enswitch?stype=internal: Resolving type '', number '00393337974334'. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Resolving type 'external', number '00393337974334'." 5 agi://127.0.0.1/enswitch?stype=internal: Resolving type 'external', number '00393337974334'. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Customer 27 has 2359 outbound seconds available." 9 agi://127.0.0.1/enswitch?stype=internal: Customer 27 has 2359 outbound seconds available. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Writing entry to active table for 1207307011.895, external:1234 - external:00393337974334." 9 agi://127.0.0.1/enswitch?stype=internal: Writing entry to active table for 1207307011.895, external:1234 - external:00393337974334. AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE TIMEOUT(absolute) 2359 -- Channel will hangup at 2008-04-04 11:42:50 UTC. AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-MUSIC default AGI Tx >> 200 result=1 AGI Rx << EXEC SetMusicOnHold default -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE __ENSWITCH-DCUSTOMER "27" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DTYPE "external" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DNUMBER "00393337974334" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-PRESENTATION "1" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-TRANSFER-CUSTOMER "27" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-UNIQUEID "1207307011.895" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-CUTOFF "2359" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DIAL-TIME "" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DIAL-OPTIONS "oL(2359000:60000)" AGI Tx >> 200 result=1 AGI Rx << EXEC SetCallerPres allowed_passed_screen -- AGI Script Executing Application: (SetCallerPres) Options: (allowed_passed_screen) AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE CALLERID(number) "1234" AGI Tx >> 200 result=1 AGI Rx << EXEC Dial Local/00393337974334@to-voipxtrade/n|600|o -- AGI Script Executing Application: (Dial) Options: (Local/00393337974334@to-voipxtrade/n|600|o) -- Called 00393337974334@to-voipxtrade/n -- Executing [00393337974334@to-voipxtrade:1] SetCallerPres("Local/00393337974334@to-voipxtrade-05d6,2", "allowed") in new stack -- Executing [00393337974334@to-voipxtrade:2] Dial("Local/00393337974334@to-voipxtrade-05d6,2", "SIP/393337974334@voipxtrade||oL(2359000:60000)") in new stack -- Limit Data for this call: > timelimit = 2359000 > play_warning = 60000 > play_to_caller = yes > play_to_callee = no > warning_freq = 0 > start_sound = (null) > warning_sound = timeleft > end_sound = (null) Audio is at 87.247.224.11 port 11004 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 216.19.176.240:5060: INVITE sip:393337974334@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39;rport From: "1234" ;tag=as3dfb9d89 To: Contact: Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 CSeq: 102 INVITE User-Agent: Integrics Enswitch Max-Forwards: 70 Date: Fri, 04 Apr 2008 11:03:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.224.11 s=session c=IN IP4 87.247.224.11 t=0 0 m=audio 11004 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 393337974334@voipxtrade asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39 From: "1234" ;tag=as3dfb9d89 Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 To: ;tag=04040308123012295979379285 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 183 Session Progress CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39 From: "1234" ;tag=as3dfb9d89 Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 To: ;tag=04040308123012295979379285 Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=VoipSwitch 10284 10284 IN IP4 216.19.176.240 s=VoipSIP i=Audio Session c=IN IP4 216.19.176.240 t=0 0 m=audio 9284 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (9 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.19.176.240:9284 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.19.176.240:9284 asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 180 Ringing CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39 From: "1234" ;tag=as3dfb9d89 Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 To: ;tag=04040308123012295979379285 Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=VoipSwitch 10284 10284 IN IP4 216.19.176.240 s=VoipSIP i=Audio Session =IN IP4 216.19.176.240 t=0 0 m=audio 9284 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (9 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.19.176.240:9284 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.19.176.240:9284 -- SIP/voipxtrade-0891b038 is making progress passing it to Local/00393337974334@to-voipxtrade-05d6,2 -- SIP/voipxtrade-0891b038 is ringing -- SIP/voipxtrade-0891b038 is making progress passing it to Local/00393337974334@to-voipxtrade-05d6,2 -- Local/00393337974334@to-voipxtrade-05d6,1 is making progress passing it to SIP/5060-088fb6e0 Audio is at 87.247.224.11 port 11848 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Record-Route: From: "1234" ;tag=3416295864-555413 To: ;tag=as7dea82ed Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.224.11 s=session c=IN IP4 87.247.224.11 t=0 0 m=audio 11848 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Local/00393337974334@to-voipxtrade-05d6,1 is ringing asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Record-Route: From: "1234" ;tag=3416295864-555413 To: ;tag=as7dea82ed Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Local/00393337974334@to-voipxtrade-05d6,1 is making progress passing it to SIP/5060-088fb6e0 [Apr 4 13:03:32] NOTICE[7635]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.19.176.240 asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> BYE sip:@87.247.224.11:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK6426.cd086d53.0 Via: SIP/2.0/UDP 87.247.224.18:5060;rport=51205;x-route-tag="tgrp:TO_DMS100" From: ;tag=D39D0AF4-23E3 To: ;tag=as32553039 Date: Fri, 04 Apr 2008 11:02:21 GMT Call-ID: 6EA0EC0E-16D11DD-88F4CC73-1113E0FD@87.247.224.18 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 5 Route: Timestamp: 1207307012 CSeq: 102 BYE Content-Length: 0 X-Enswitch-Source: 87.247.224.18:51205 X-Enswitch-External: yes <-------------> --- (16 headers 0 lines) --- Sending to 87.247.224.5 : 5060 (NAT) <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK6426.cd086d53.0;received=87.247.224.5 Via: SIP/2.0/UDP 87.247.224.18:5060;rport=51205;x-route-tag="tgrp:TO_DMS100" Record-Route: From: ;tag=D39D0AF4-23E3 To: ;tag=as32553039 Call-ID: 6EA0EC0E-16D11DD-88F4CC73-1113E0FD@87.247.224.18 CSeq: 102 BYE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Apr 4 13:03:35] WARNING[7635]: rtp.c:891 ast_rtcp_read: RTCP Read too short Scheduling destruction of SIP dialog '7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 212.177.111.212:5060: CANCEL sip:00983116684488@212.177.111.212 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5bbd73d0;rport From: "0793079899" ;tag=as5fb7c6fd To: Call-ID: 7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11 CSeq: 102 CANCEL User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11' in 32000 ms (Method: INVITE) AGI Tx >> 200 result=-1 == Spawn extension (from-internal, 800098420, 1) exited non-zero on 'SIP/87.247.224.18-088fa150' -- Executing [h@from-internal:1] DeadAGI("SIP/87.247.224.18-088fa150", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end == Spawn extension (to-verizon, 00983116684488, 2) exited non-zero on 'Local/00983116684488@to-verizon-3a0a,2' -- Executing [h@to-verizon:1] DeadAGI("Local/00983116684488@to-verizon-3a0a,2", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: SIP/87.247.224.18-088fa150 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1207306944.889 AGI Tx >> agi_callerid: 079276429 AGI Tx >> agi_calleridname: unknown AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 800098420 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from-internal AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: Local/00983116684488@to-verizon-3a0a,2 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: Local AGI Tx >> agi_uniqueid: 1207306991.893 AGI Tx >> agi_callerid: 0793079899 AGI Tx >> agi_calleridname: unknown AGI Tx >> agi_callingpres: 3 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: unknown AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: to-verizon AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Tx >> 200 result=-1 AGI Rx << GET VARIABLE CHANLOCALSTATUS AGI Tx >> 200 result=0 AGI Rx << VERBOSE "LOCAL STATUS = " 0 agi://127.0.0.1/enswitch?scustomer=17&stype=card&snumber=208896120293&ctype=external&cnumber=00983116684488&loopback=1: LOCAL STATUS = AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) -- AGI Script agi://127.0.0.1/enswitch?scustomer=17&stype=card&snumber=208896120293&ctype=external&cnumber=00983116684488&loopback=1 completed, returning 0 -- Executing [h@enswitch-call:1] DeadAGI("Local/00983116684488@enswitch-call-7290,2", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: Local/00983116684488@enswitch-call-7290,2 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: Local AGI Tx >> agi_uniqueid: 1207306991.891 AGI Tx >> agi_callerid: 0793079899 AGI Tx >> agi_calleridname: unknown AGI Tx >> agi_callingpres: 1 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: unknown AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: enswitch-call AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207306944.889) AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207306944.889) AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207306944.889) AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 -- AGI Script agi://127.0.0.1/end completed, returning 0 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,rtt) [Apr 4 13:03:35] WARNING[7630]: func_channel.c:96 func_channel_read: Unknown or unavailable item requested: 'rtpqos,audio,rtt' AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_lostpackets) [Apr 4 13:03:35] WARNING[7630]: func_channel.c:96 func_channel_read: Unknown or unavailable item requested: 'rtpqos,audio,local_lostpackets' AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_jitter) [Apr 4 13:03:35] WARNING[7630]: func_channel.c:96 func_channel_read: Unknown or unavailable item requested: 'rtpqos,audio,local_jitter' AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_lostpackets) [Apr 4 13:03:35] WARNING[7630]: func_channel.c:96 func_channel_read: Unknown or unavailable item requested: 'rtpqos,audio,remote_lostpackets' AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_jitter) [Apr 4 13:03:35] WARNING[7630]: func_channel.c:96 func_channel_read: Unknown or unavailable item requested: 'rtpqos,audio,remote_jitter' AGI Tx >> 200 result=0 asterisk2*CLI> <--- SIP read from 212.177.111.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5bbd73d0;rport From: "0793079899" ;tag=as5fb7c6fd To: ;tag=z07m5db13cb-cbs720058743-i-1483 Call-ID: 7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11 CSeq: 102 CANCEL Supported: replaces User-agent: Patton SN4960 4E30V UI MxSF v3.2.8.45 00A0BA02AA67 R4.2 2007-12-03 H323 RBS SIP Content-Length: 0 <-------------> --- (9 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 212.177.111.212:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5bbd73d0;rport From: "0793079899" ;tag=as5fb7c6fd To: ;tag=z07m5db13cb-cbs720058743-i-1483 Call-ID: 7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11 CSeq: 102 INVITE User-agent: Patton SN4960 4E30V UI MxSF v3.2.8.45 00A0BA02AA67 R4.2 2007-12-03 H323 RBS SIP Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 212.177.111.212:5060: ACK sip:00983116684488@212.177.111.212 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5bbd73d0;rport From: "0793079899" ;tag=as5fb7c6fd To: ;tag=z07m5db13cb-cbs720058743-i-1483 Contact: Call-ID: 7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11 CSeq: 102 ACK User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '7c1c2ed146cc34a8691f9dde3ef040b6@87.247.224.11' Method: INVITE AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,rtt) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_jitter) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_jitter) AGI Tx >> 200 result=1 (CANCEL) -- AGI Script agi://127.0.0.1/end completed, returning 0 asterisk2*CLI> <--- SIP read from 87.247.224.6:46178 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46177;branch=z9hG4bK.3c22a392;rport;alias From: sip:sipsak@87.247.224.6:46177;tag=2a2f85b6 To: sip:87.247.224.11:5060 Call-ID: 707757494@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46177 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.6:46178 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46177;branch=z9hG4bK.3c22a392;alias;received=87.247.224.6;rport=46178 From: sip:sipsak@87.247.224.6:46177;tag=2a2f85b6 To: sip:87.247.224.11:5060;tag=as77da15fa Call-ID: 707757494@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 ------------>> Scheduling destruction of SIP dialog '707757494@87.247.224.6' in 32000 ms (Method: OPTIONS) -- AGI Script agi://127.0.0.1/end completed, returning 0 Really destroying SIP dialog '6EA0EC0E-16D11DD-88F4CC73-1113E0FD@87.247.224.18' Method: BYE [Apr 4 13:03:37] WARNING[7635]: rtp.c:891 ast_rtcp_read: RTCP Read too short Really destroying SIP dialog '392867653@87.247.224.6' Method: OPTIONS [Apr 4 13:03:42] WARNING[7635]: rtp.c:891 ast_rtcp_read: RTCP Read too short asterisk2*CLI> <--- SIP read from 87.247.224.6:46180 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46179;branch=z9hG4bK.1e594212;rport;alias From: sip:sipsak@87.247.224.6:46179;tag=2865590c To: sip:87.247.224.11:5060 Call-ID: 677730572@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46179 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46180 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46179;branch=z9hG4bK.1e594212;alias;received=87.247.224.6;rport=46180 From: sip:sipsak@87.247.224.6:46179;tag=2865590c To: sip:87.247.224.11:5060;tag=as0765c294 Call-ID: 677730572@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '677730572@87.247.224.6' in 32000 ms (Method: OPTIONS) [Apr 4 13:03:45] WARNING[7635]: rtp.c:891 ast_rtcp_read: RTCP Read too short Really destroying SIP dialog '906843159@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> CANCEL sip:00393337974334@87.247.224.11:5060 SIP/2.0 Record-Route: Max-Forwards: 68 To: From: "1234" ;tag=3416295864-555413 Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Contact: Content-Length: 0 X-Enswitch-Source: 213.166.9.34:5060 X-Enswitch-External: yes <-------------> --- (14 headers 0 lines) --- Sending to 87.247.224.5 : 5060 (NAT) <--- Reliably Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f From: "1234" ;tag=3416295864-555413 To: ;tag=as7dea82ed Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK26bd155adce2ef48a3ba6633bb87608f Record-Route: From: "1234" ;tag=3416295864-555413 To: ;tag=as7dea82ed Call-ID: 2644-3416295864-555405@MSX2.mydomain.com CSeq: 1 CANCEL User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> ACK sip:00393337974334@87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK0229.1f7e6805.0 From: "1234" ;tag=3416295864-555413 Call-ID: 2644-3416295864-555405@MSX2.mydomain.com To: ;tag=as7dea82ed CSeq: 1 ACK User-Agent: Enswitch SIP proxy Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '18f7857467025e7e4eae9b1934078a4d@87.247.224.11' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 216.19.176.240:5060: CANCEL sip:393337974334@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39;rport From: "1234" ;tag=as3dfb9d89 To: Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 CSeq: 102 CANCEL User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '18f7857467025e7e4eae9b1934078a4d@87.247.224.11' in 32000 ms (Method: INVITE) AGI Tx >> 200 result=-1 == Spawn extension (to-voipxtrade, 00393337974334, 2) exited non-zero on 'Local/00393337974334@to-voipxtrade-05d6,2' == Spawn extension (from-internal, 00393337974334, 1) exited non-zero on 'SIP/5060-088fb6e0' -- Executing [h@from-internal:1] DeadAGI("SIP/5060-088fb6e0", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: SIP/5060-088fb6e0 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1207307011.895 AGI Tx >> agi_callerid: 1234 AGI Tx >> agi_calleridname: 1234 AGI Tx >> agi_callingpres: 1 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 00393337974334 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from-internal AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> -- Executing [h@to-voipxtrade:1] DeadAGI("Local/00393337974334@to-voipxtrade-05d6,2", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: Local/00393337974334@to-voipxtrade-05d6,2 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: Local AGI Tx >> agi_uniqueid: 1207307011.897 AGI Tx >> agi_callerid: 1234 AGI Tx >> agi_calleridname: 1234 AGI Tx >> agi_callingpres: 3 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: unknown AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: to-voipxtrade AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> CLI> AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207307011.895) AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207307011.895) AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 -- AGI Script agi://127.0.0.1/end completed, returning 0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,rtt) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_jitter) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_jitter) AGI Tx >> 200 result=1 (CANCEL) asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 200 OK CSeq: 102 CANCEL Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39 From: "1234" ;tag=as3dfb9d89 Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 To: ;tag=04040308123012295979379285 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 487 Request Terminated CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39 From: "1234" ;tag=as3dfb9d89 Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 To: ;tag=04040308123012295979379285 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 216.19.176.240:5060: ACK sip:393337974334@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK5de47a39;rport From: "1234" ;tag=as3dfb9d89 To: ;tag=04040308123012295979379285 Contact: Call-ID: 18f7857467025e7e4eae9b1934078a4d@87.247.224.11 CSeq: 102 ACK User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '18f7857467025e7e4eae9b1934078a4d@87.247.224.11' Method: INVITE -- AGI Script agi://127.0.0.1/end completed, returning 0 Really destroying SIP dialog '2644-3416295864-555405@MSX2.mydomain.com' Method: ACK asterisk2*CLI> <--- SIP read from 87.247.224.6:46182 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46181;branch=z9hG4bK.23334379;rport;alias From: sip:sipsak@87.247.224.6:46181;tag=67715fab To: sip:87.247.224.11:5060 Call-ID: 1735483307@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46181 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.6:46182 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46181;branch=z9hG4bK.23334379;alias;received=87.247.224.6;rport=46182 From: sip:sipsak@87.247.224.6:46181;tag=67715fab To: sip:87.247.224.11:5060;tag=as0cf47859 Call-ID: 1735483307@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1735483307@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1539445031@87.247.224.6' Method: OPTIONS == Parsing '/etc/asterisk/manager.conf': Found AGI Rx << SET VARIABLE TIMEOUT(absolute) 86400 -- Channel will hangup at 2008-04-05 11:09:01 UTC. AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-SCUSTOMER AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(X-Enswitch-Source) AGI Tx >> 200 result=1 (213.166.9.34:5060) AGI Rx << GET VARIABLE ENSWITCH-PRESENTATION AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE SIP_HEADER(X-Enswitch-External) AGI Tx >> 200 result=1 (yes) AGI Rx << GET VARIABLE ENSWITCH-TRANSFER-PHONE AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-TRANSFER-CUSTOMER AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Resolving type '', number '00393397069435'." 5 agi://127.0.0.1/enswitch?stype=internal: Resolving type '', number '00393397069435'. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Resolving type 'external', number '00393397069435'." 5 agi://127.0.0.1/enswitch?stype=internal: Resolving type 'external', number '00393397069435'. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Customer 27 has 2348 outbound seconds available." 9 agi://127.0.0.1/enswitch?stype=internal: Customer 27 has 2348 outbound seconds available. AGI Tx >> 200 result=1 AGI Rx << VERBOSE "Writing entry to active table for 1207307341.909, external:1234 - external:00393397069435." 9 agi://127.0.0.1/enswitch?stype=internal: Writing entry to active table for 1207307341.909, external:1234 - external:00393397069435. AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE TIMEOUT(absolute) 2348 -- Channel will hangup at 2008-04-04 11:48:09 UTC. AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-MUSIC default AGI Tx >> 200 result=1 AGI Rx << EXEC SetMusicOnHold default -- AGI Script Executing Application: (SetMusicOnHold) Options: (default) AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE __ENSWITCH-DCUSTOMER "27" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DTYPE "external" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DNUMBER "00393397069435" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-PRESENTATION "1" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-TRANSFER-CUSTOMER "27" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-UNIQUEID "1207307341.909" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-CUTOFF "2348" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DIAL-TIME "" AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE __ENSWITCH-DIAL-OPTIONS "oL(2348000:60000)" AGI Tx >> 200 result=1 AGI Rx << EXEC SetCallerPres allowed_passed_screen -- AGI Script Executing Application: (SetCallerPres) Options: (allowed_passed_screen) AGI Tx >> 200 result=0 AGI Rx << SET VARIABLE CALLERID(number) "1234" AGI Tx >> 200 result=1 AGI Rx << EXEC Dial Local/00393397069435@to-voipxtrade/n|600|o -- AGI Script Executing Application: (Dial) Options: (Local/00393397069435@to-voipxtrade/n|600|o) -- Called 00393397069435@to-voipxtrade/n -- Executing [00393397069435@to-voipxtrade:1] SetCallerPres("Local/00393397069435@to-voipxtrade-c68a,2", "allowed") in new stack -- Executing [00393397069435@to-voipxtrade:2] Dial("Local/00393397069435@to-voipxtrade-c68a,2", "SIP/393397069435@voipxtrade||oL(2348000:60000)") in new stack -- Limit Data for this call: > timelimit = 2348000 > play_warning = 60000 > play_to_caller = yes > play_to_callee = no > warning_freq = 0 > start_sound = (null) > warning_sound = timeleft > end_sound = (null) Audio is at 87.247.224.11 port 13316 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 216.19.176.240:5060: INVITE sip:393397069435@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093;rport From: "1234" ;tag=as69f23221 To: Contact: Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 CSeq: 102 INVITE User-Agent: Integrics Enswitch Max-Forwards: 70 Date: Fri, 04 Apr 2008 11:09:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.224.11 s=session c=IN IP4 87.247.224.11 t=0 0 m=audio 13316 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 393397069435@voipxtrade asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093 From: "1234" ;tag=as69f23221 Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 To: ;tag=04040908120012299275629341 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 183 Session Progress CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093 From: "1234" ;tag=as69f23221 Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 To: ;tag=04040908120012299275629341 Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=VoipSwitch 10340 10340 IN IP4 216.19.176.240 s=VoipSIP i=Audio Session c=IN IP4 216.19.176.240 t=0 0 m=audio 9340 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (9 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.19.176.240:9340 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 216.19.176.240:9340 <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 180 Ringing CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093 From: "1234" ;tag=as69f23221 Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 To: ;tag=04040908120012299275629341 Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=VoipSwitch 10340 10340 IN IP4 216.19.176.240 s=VoipSIP i=Audio Session c=IN IP4 216.19.176.240 t=0 0 m=audio 9340 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (9 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 216.19.176.240:9340 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/voipxtrade-088f1f78 is making progress passing it to Local/00393397069435@to-voipxtrade-c68a,2 Peer audio RTP is at port 216.19.176.240:9340 -- SIP/voipxtrade-088f1f78 is ringing -- SIP/voipxtrade-088f1f78 is making progress passing it to Local/00393397069435@to-voipxtrade-c68a,2 -- Local/00393397069435@to-voipxtrade-c68a,1 is making progress passing it to SIP/5060-088edb90 Audio is at 87.247.224.11 port 16332 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK2125cde8ba44f6404974f20a09430994 Record-Route: From: "1234" ;tag=3416296194-177369 To: ;tag=as04baa98a Call-ID: 2687-3416296194-177360@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.224.11 s=session c=IN IP4 87.247.224.11 t=0 0 m=audio 16332 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Local/00393397069435@to-voipxtrade-c68a,1 is ringing asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK2125cde8ba44f6404974f20a09430994 Record-Route: From: "1234" ;tag=3416296194-177369 To: ;tag=as04baa98a Call-ID: 2687-3416296194-177360@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Local/00393397069435@to-voipxtrade-c68a,1 is making progress passing it to SIP/5060-088edb90 [Apr 4 13:09:03] NOTICE[7678]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 216.19.176.240 [Apr 4 13:09:04] WARNING[7678]: rtp.c:891 ast_rtcp_read: RTCP Read too short asterisk2*CLI> <--- SIP read from 87.247.224.6:46254 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46253;branch=z9hG4bK.2ff01609;rport;alias From: sip:sipsak@87.247.224.6:46253;tag=1debe01c To: sip:87.247.224.11:5060 Call-ID: 501997596@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46253 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46254 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46253;branch=z9hG4bK.2ff01609;alias;received=87.247.224.6;rport=46254 From: sip:sipsak@87.247.224.6:46253;tag=1debe01c To: sip:87.247.224.11:5060;tag=as38f623e8 Call-ID: 501997596@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1115517465@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1465466963@87.247.224.6' Method: OPTIONS [Apr 4 13:09:29] WARNING[7678]: rtp.c:891 ast_rtcp_read: RTCP Read too short asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> CANCEL sip:00393397069435@87.247.224.11:5060 SIP/2.0 Record-Route: Max-Forwards: 68 To: From: "1234" ;tag=3416296194-177369 Call-ID: 2687-3416296194-177360@MSX2.mydomain.com CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK2125cde8ba44f6404974f20a09430994 Contact: Content-Length: 0 X-Enswitch-Source: 213.166.9.34:5060 X-Enswitch-External: yes <-------------> --- (14 headers 0 lines) --- Sending to 87.247.224.5 : 5060 (NAT) <--- Reliably Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK2125cde8ba44f6404974f20a09430994 From: "1234" ;tag=3416296194-177369 To: ;tag=as04baa98a Call-ID: 2687-3416296194-177360@MSX2.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0;received=87.247.224.5 Via: SIP/2.0/UDP 213.166.9.34:5060;branch=z9hG4bK2125cde8ba44f6404974f20a09430994 Record-Route: From: "1234" ;tag=3416296194-177369 To: ;tag=as04baa98a Call-ID: 2687-3416296194-177360@MSX2.mydomain.com CSeq: 1 CANCEL User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> asterisk2*CLI> <--- SIP read from 87.247.224.5:5060 ---> ACK sip:00393397069435@87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.5;branch=z9hG4bK5f3d.acea40f.0 From: "1234" ;tag=3416296194-177369 Call-ID: 2687-3416296194-177360@MSX2.mydomain.com To: ;tag=as04baa98a CSeq: 1 ACK User-Agent: Enswitch SIP proxy Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Apr 4 13:09:31] WARNING[7676]: app_dial.c:671 wait_for_answer: Unable to forward voice frame Scheduling destruction of SIP dialog '1fab15a63830e0a55c39a16649c0f571@87.247.224.11' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 216.19.176.240:5060: CANCEL sip:393397069435@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093;rport From: "1234" ;tag=as69f23221 To: Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 CSeq: 102 CANCEL User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '1fab15a63830e0a55c39a16649c0f571@87.247.224.11' in 32000 ms (Method: INVITE) AGI Tx >> 200 result=-1 == Spawn extension (from-internal, 00393397069435, 1) exited non-zero on 'SIP/5060-088edb90' -- Executing [h@from-internal:1] DeadAGI("SIP/5060-088edb90", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: SIP/5060-088edb90 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1207307341.909 AGI Tx >> agi_callerid: 1234 AGI Tx >> agi_calleridname: 1234 AGI Tx >> agi_callingpres: 1 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: 00393397069435 AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from-internal AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> CLI> == Spawn extension (to-voipxtrade, 00393397069435, 2) exited non-zero on 'Local/00393397069435@to-voipxtrade-c68a,2' -- Executing [h@to-voipxtrade:1] DeadAGI("Local/00393397069435@to-voipxtrade-c68a,2", "agi://127.0.0.1/end") in new stack AGI Tx >> agi_network: yes AGI Tx >> agi_network_script: end AGI Tx >> agi_request: agi://127.0.0.1/end AGI Tx >> agi_channel: Local/00393397069435@to-voipxtrade-c68a,2 AGI Tx >> agi_language: it_IT AGI Tx >> agi_type: Local AGI Tx >> agi_uniqueid: 1207307341.911 AGI Tx >> agi_callerid: 1234 AGI Tx >> agi_calleridname: 1234 AGI Tx >> agi_callingpres: 3 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: unknown AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: to-voipxtrade AGI Tx >> agi_extension: h AGI Tx >> agi_priority: 1 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << SET VARIABLE LANGUAGE() it_IT AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207307341.909) AGI Rx << GET VARIABLE ENSWITCH-CALLID AGI Tx >> 200 result=1 (1207307341.909) AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << HANGUP AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ANSWEREDTIME AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE DIALSTATUS AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 AGI Rx << GET VARIABLE ENSWITCH-FAX-FILE AGI Tx >> 200 result=0 -- AGI Script agi://127.0.0.1/end completed, returning 0 AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,rtt) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,local_jitter) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_lostpackets) AGI Tx >> 200 result=1 (CANCEL) AGI Rx << GET VARIABLE CHANNEL(rtpqos,audio,remote_jitter) AGI Tx >> 200 result=1 (CANCEL) asterisk2*CLI> <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 200 OK CSeq: 102 CANCEL Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093 From: "1234" ;tag=as69f23221 Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 To: ;tag=04040908120012299275629341 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 216.19.176.240:5060 ---> SIP/2.0 487 Request Terminated CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093 From: "1234" ;tag=as69f23221 Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 To: ;tag=04040908120012299275629341 Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 216.19.176.240:5060: ACK sip:393397069435@216.19.176.240 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.11:5060;branch=z9hG4bK65187093;rport From: "1234" ;tag=as69f23221 To: ;tag=04040908120012299275629341 Contact: Call-ID: 1fab15a63830e0a55c39a16649c0f571@87.247.224.11 CSeq: 102 ACK User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '1fab15a63830e0a55c39a16649c0f571@87.247.224.11' Method: INVITE -- AGI Script agi://127.0.0.1/end completed, returning 0 Really destroying SIP dialog '2687-3416296194-177360@MSX2.mydomain.com' Method: ACK asterisk2*CLI> <--- SIP read from 87.247.224.6:46261 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46260;branch=z9hG4bK.06651815;rport;alias From: sip:sipsak@87.247.224.6:46260;tag=56e633a9 To: sip:87.247.224.11:5060 Call-ID: 1457927081@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46260 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46261 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46260;branch=z9hG4bK.06651815;alias;received=87.247.224.6;rport=46261 From: sip:sipsak@87.247.224.6:46260;tag=56e633a9 To: sip:87.247.224.11:5060;tag=as357d109b Call-ID: 1457927081@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1457927081@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '501997596@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.6:46264 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46263;branch=z9hG4bK.34155717;rport;alias From: sip:sipsak@87.247.224.6:46263;tag=b36d679 To: sip:87.247.224.11:5060 Call-ID: 188143225@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46263 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46264 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46263;branch=z9hG4bK.34155717;alias;received=87.247.224.6;rport=46264 From: sip:sipsak@87.247.224.6:46263;tag=b36d679 To: sip:87.247.224.11:5060;tag=as43e1f603 Call-ID: 188143225@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '188143225@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1220304236@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.6:46266 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46265;branch=z9hG4bK.5ca44218;rport;alias From: sip:sipsak@87.247.224.6:46265;tag=69fd80e0 To: sip:87.247.224.11:5060 Call-ID: 1778221280@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46265 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46266 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46265;branch=z9hG4bK.5ca44218;alias;received=87.247.224.6;rport=46266 From: sip:sipsak@87.247.224.6:46265;tag=69fd80e0 To: sip:87.247.224.11:5060;tag=as13ad4608 Call-ID: 1778221280@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1778221280@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1115517465@87.247.224.6' Method: OPTIONS == Parsing '/etc/asterisk/manager.conf': Found asterisk2*CLI> <--- SIP read from 87.247.224.6:46268 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46267;branch=z9hG4bK.6302e0ec;rport;alias From: sip:sipsak@87.247.224.6:46267;tag=3f179ba3 To: sip:87.247.224.11:5060 Call-ID: 1058511779@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46267 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46268 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46267;branch=z9hG4bK.6302e0ec;alias;received=87.247.224.6;rport=46268 From: sip:sipsak@87.247.224.6:46267;tag=3f179ba3 To: sip:87.247.224.11:5060;tag=as5cd4d224 Call-ID: 1058511779@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1058511779@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1457927081@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.6:46270 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46269;branch=z9hG4bK.69b4a6a2;rport;alias From: sip:sipsak@87.247.224.6:46269;tag=45583e60 To: sip:87.247.224.11:5060 Call-ID: 1163411040@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46269 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) <--- Transmitting (NAT) to 87.247.224.6:46270 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46269;branch=z9hG4bK.69b4a6a2;alias;received=87.247.224.6;rport=46270 From: sip:sipsak@87.247.224.6:46269;tag=45583e60 To: sip:87.247.224.11:5060;tag=as10b3f709 Call-ID: 1163411040@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1163411040@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '188143225@87.247.224.6' Method: OPTIONS asterisk2*CLI> <--- SIP read from 87.247.224.6:46272 ---> OPTIONS sip:87.247.224.11:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.224.6:46271;branch=z9hG4bK.58a16b31;rport;alias From: sip:sipsak@87.247.224.6:46271;tag=535e2369 To: sip:87.247.224.11:5060 Call-ID: 1398678377@87.247.224.6 CSeq: 1 OPTIONS Contact: sip:sipsak@87.247.224.6:46271 Content-Length: 0 Max-Forwards: 70 User-Agent: sipsak 0.9.6 Accept: text/plain <-------------> --- (11 headers 0 lines) --- Looking for s in from-external (domain 87.247.224.11) asterisk2*CLI> <--- Transmitting (NAT) to 87.247.224.6:46272 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.224.6:46271;branch=z9hG4bK.58a16b31;alias;received=87.247.224.6;rport=46272 From: sip:sipsak@87.247.224.6:46271;tag=535e2369 To: sip:87.247.224.11:5060;tag=as37c8a3b3 Call-ID: 1398678377@87.247.224.6 CSeq: 1 OPTIONS User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '1398678377@87.247.224.6' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog '1778221280@87.247.224.6' Method: OPTIONS asterisk2*CLI>