[2011-02-09 09:55:57] VERBOSE[1280] logger.c: Reliably Transmitting (no NAT) to 192.168.125.148:5060: OPTIONS sip:ctvsh_1303@192.168.125.148;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK53b20c4f;rport From: "asterisk" ;tag=as46079706 To: Contact: Call-ID: 587de21e1b90d16a42954be51af45f4d@174.47.138.162 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 09 Feb 2011 15:55:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.125.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK53b20c4f;rport From: "asterisk" ;tag=as46079706 To: ;tag=72E49A00-F6D2DD19 CSeq: 102 OPTIONS Call-ID: 587de21e1b90d16a42954be51af45f4d@174.47.138.162 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Supported: 100rel,timer,replaces Content-Length: 0 <-------------> <--- SIP read from 192.168.125.148:5060 ---> INVITE sip:*724375@pbx01.itfreedom.com:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bK94c426936C24E374 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: CSeq: 1 INVITE Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 300 v=0 o=- 1297245613 1297245613 IN IP4 192.168.125.148 s=Polycom IP Phone c=IN IP4 192.168.125.148 t=0 0 a=sendrecv m=audio 2258 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> [2011-02-09 09:55:57] VERBOSE[1280] logger.c: --- (15 headers 13 lines) --- [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Sending to 192.168.125.148 : 5060 (no NAT) [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Using INVITE request as basis request - 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.125.148:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bK94c426936C24E374;received=192.168.125.148 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as6ffd1ff8 Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7befc7a0" Content-Length: 0 <------------> [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found user 'ctvsh_1303_3000' [2011-02-09 09:55:57] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.148:5060 ---> ACK sip:*724375@pbx01.itfreedom.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bK94c426936C24E374 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as6ffd1ff8 CSeq: 1 ACK Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 Content-Length: 0 --- [2011-02-09 09:55:57] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.148:5060 ---> INVITE sip:*724375@pbx01.itfreedom.com:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bKb73d6872B37ACE3B From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: CSeq: 2 INVITE Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="ctvsh_1303_3000", realm="asterisk", nonce="7befc7a0", uri="sip:*724375@pbx01.itfreedom.com:5060;user=phone;transport=udp", response="5840884df8af64155e3cb1f235437598", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 300 v=0 o=- 1297245613 1297245613 IN IP4 192.168.125.148 s=Polycom IP Phone c=IN IP4 192.168.125.148 t=0 0 a=sendrecv m=audio 2258 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> [2011-02-09 09:55:57] VERBOSE[1280] logger.c: --- (16 headers 13 lines) --- [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Sending to 192.168.125.148 : 5060 (no NAT) [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Using INVITE request as basis request - 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found user 'ctvsh_1303_3000' [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found RTP audio format 9 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found RTP audio format 0 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found RTP audio format 8 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found RTP audio format 18 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found RTP audio format 101 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found audio description format G722 for ID 9 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found audio description format PCMU for ID 0 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found audio description format PCMA for ID 8 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found audio description format G729 for ID 18 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Found audio description format telephone-event for ID 101 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Peer audio RTP is at port 192.168.125.148:2258 [2011-02-09 09:55:57] VERBOSE[1280] logger.c: Looking for *724375 in ctvsh_north-internal (domain pbx01.itfreedom.com) [2011-02-09 09:55:57] VERBOSE[1280] logger.c: list_route: hop: [2011-02-09 09:55:57] VERBOSE[1280] logger.c: <--- Transmitting (no NAT) to 192.168.125.148:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bKb73d6872B37ACE3B;received=192.168.125.148 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [*724375@ctvsh_north-internal:1] Goto("SIP/ctvsh_1303_3000-000008de", "ctvsh-page|1|1") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Goto (ctvsh-page,1,1) [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [1@ctvsh-page:1] Macro("SIP/ctvsh_1303_3000-000008de", "system-page-with-recording|"Client Care North" <1303>|ctvsh_north|2|2") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [s@macro-system-page-with-recording:1] Set("SIP/ctvsh_1303_3000-000008de", "callerid="Client Care North" <1303>") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [s@macro-system-page-with-recording:2] Set("SIP/ctvsh_1303_3000-000008de", "zone=ctvsh_north") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [s@macro-system-page-with-recording:3] Set("SIP/ctvsh_1303_3000-000008de", "num-times=2") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Goto (macro-system-page-with-recording,do-page,1) [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [do-page@macro-system-page-with-recording:1] Set("SIP/ctvsh_1303_3000-000008de", "page-recording=tmp/ctvsh_north-page-message-456011.wav") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [do-page@macro-system-page-with-recording:2] NoOp("SIP/ctvsh_1303_3000-000008de", "Client Care North <1303> is paging a recorded message 2 time(s) to zone ctvsh_north...") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [do-page@macro-system-page-with-recording:3] Macro("SIP/ctvsh_1303_3000-000008de", "system-page-fetch-regs|Client Care North <1303>|ctvsh_north") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Executing [do-page@macro-system-page-with-recording:4] Record("SIP/ctvsh_1303_3000-000008de", "tmp/ctvsh_north-page-message-456011.wav") in new stack [2011-02-09 09:55:57] VERBOSE[3274] logger.c: Audio is at 174.47.138.162 port 14346 [2011-02-09 09:55:57] VERBOSE[3274] logger.c: Adding codec 0x4 (ulaw) to SDP [2011-02-09 09:55:57] VERBOSE[3274] logger.c: Adding codec 0x8 (alaw) to SDP [2011-02-09 09:55:57] VERBOSE[3274] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-09 09:55:57] VERBOSE[3274] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.125.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bKb73d6872B37ACE3B;received=192.168.125.148 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as424636ea Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 1268 1268 IN IP4 174.47.138.162 s=session c=IN IP4 174.47.138.162 t=0 0 m=audio 14346 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2011-02-09 09:55:57] VERBOSE[3274] logger.c: -- Playing 'beep' (language 'en') [2011-02-09 09:55:57] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.148:5060 ---> ACK sip:*724375@174.47.138.162;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bKbcc78fa67AE9724F From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as424636ea CSeq: 2 ACK Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> [2011-02-09 09:56:00] VERBOSE[1280] logger.c: --- (14 headers 0 lines) --- [2011-02-09 09:56:00] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.148:5060 ---> BYE sip:*724375@174.47.138.162;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bK8b1438d035449F69 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as424636ea CSeq: 3 BYE Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.1.3.0439 Accept-Language: en Proxy-Authorization: Digest username="ctvsh_1303_3000", realm="asterisk", nonce="7befc7a0", uri="sip:*724375@pbx01.itfreedom.com:5060;user=phone;transport=udp", response="ad6c02c46eff934c26fe9c7309cc279b", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-02-09 09:56:00] VERBOSE[1280] logger.c: --- (12 headers 0 lines) --- [2011-02-09 09:56:00] VERBOSE[1280] logger.c: Sending to 192.168.125.148 : 5060 (no NAT) [2011-02-09 09:56:00] VERBOSE[1280] logger.c: <--- Transmitting (no NAT) to 192.168.125.148:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.148;branch=z9hG4bK8b1438d035449F69;received=192.168.125.148 From: "ctvsh_1303_3000" ;tag=9F16B668-B6D14341 To: ;tag=as424636ea Call-ID: 6fd3d96d-6ae2c2be-dd35cfa7@192.168.125.148 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [2011-02-09 09:56:00] VERBOSE[3274] logger.c: == Spawn extension (ctvsh-page, 1, 1) exited non-zero on 'SIP/ctvsh_1303_3000-000008de' [2011-02-09 09:56:00] VERBOSE[3274] logger.c: -- Executing [h@macro-system-page-with-recording:2] DeadAGI("SIP/ctvsh_1303_3000-000008de", "pagefile|Client Care North <1303>|tmp/ctvsh_north-page-message-456011.wav|2|2|ctvsh_north|SIP/ctvsh_1345&SIP/ctvsh_1346&SIP/ctvsh_1365&SIP/ctvsh_1367&SIP/ctvsh_9999") in new stack [2011-02-09 09:56:00] VERBOSE[3274] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/pagefile [2011-02-09 09:56:00] VERBOSE[3274] logger.c: -- AGI Script pagefile completed, returning 0 [2011-02-09 09:56:00] VERBOSE[3274] logger.c: -- Executing [h@macro-system-page-with-recording:3] NoOp("SIP/ctvsh_1303_3000-000008de", "Exiting page...") in new stack [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.175:5060: INVITE sip:ctvsh_1345@192.168.125.175;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK21e3f48d;rport From: "Client Care North" ;tag=as5910996d To: Contact: Call-ID: 785f5a0e0ee454236e844e13301ca403@174.47.138.162 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 09 Feb 2011 15:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Alert-Info: Ring Answer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1268 1268 IN IP4 174.47.138.162 s=session c=IN IP4 174.47.138.162 t=0 0 m=audio 10488 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2011-02-09 09:56:01] VERBOSE[3293] logger.c: -- Called ctvsh_1345 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Audio is at 174.47.138.162 port 15802 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x4 (ulaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x8 (alaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x2 (gsm) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.176:5060: INVITE sip:ctvsh_1346@192.168.125.176;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK701bba97;rport From: "Client Care North" ;tag=as4ce7104a To: Contact: Call-ID: 130d41e27e4785013f4c42bc27084757@174.47.138.162 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 09 Feb 2011 15:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Alert-Info: Ring Answer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1268 1268 IN IP4 174.47.138.162 s=session c=IN IP4 174.47.138.162 t=0 0 m=audio 15802 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2011-02-09 09:56:01] VERBOSE[3293] logger.c: -- Called ctvsh_1346 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Audio is at 174.47.138.162 port 14256 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x4 (ulaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x8 (alaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x2 (gsm) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.146:5060: INVITE sip:ctvsh_1365@192.168.125.146;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK652e87c5;rport From: "Client Care North" ;tag=as6caf1df2 To: Contact: Call-ID: 23c75d4a35e1201669f0297576c065d6@174.47.138.162 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 09 Feb 2011 15:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Alert-Info: Ring Answer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1268 1268 IN IP4 174.47.138.162 s=session c=IN IP4 174.47.138.162 t=0 0 m=audio 14256 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2011-02-09 09:56:01] VERBOSE[3293] logger.c: -- Called ctvsh_1365 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Audio is at 174.47.138.162 port 10836 [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x4 (ulaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x8 (alaw) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding codec 0x2 (gsm) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [2011-02-09 09:56:01] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.180:5060: INVITE sip:ctvsh_1367@192.168.125.180;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK6014faee;rport From: "Client Care North" ;tag=as02276705 To: Contact: Call-ID: 444209da526f6a966e87fd5c06a2e48d@174.47.138.162 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 09 Feb 2011 15:56:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Alert-Info: Ring Answer -- Supported: replaces Content-Length: 0 --- [2011-02-09 09:56:01] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.180:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK6014faee;rport From: "Client Care North" ;tag=as02276705 To: ;tag=E00854E7-3259039C CSeq: 102 INVITE Call-ID: 444209da526f6a966e87fd5c06a2e48d@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.1.3.0507 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 <-------------> [2011-02-09 09:56:01] VERBOSE[1280] logger.c: --- (14 headers 0 lines) --- [2011-02-09 09:56:01] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.146:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK652e87c5;rport From: "Client Care North" ;tag=as6caf1df2 To: ;tag=4E4FBDDD-29302AD0 CSeq: 102 INVITE Call-ID: 23c75d4a35e1201669f0297576c065d6@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.1.3.0507 Allow-Events: talk,hold,conference Accept-Language: en Content-Length: 0 --- [2011-02-09 09:56:01] VERBOSE[3293] logger.c: -- Created MeetMe conference 1021 for conference '870915922d' [2011-02-09 09:56:01] VERBOSE[3292] logger.c: -- Playing 'silence/1' (language 'en') [2011-02-09 09:56:01] WARNING[3292] file.c: Unexpected control subclass '-1' [2011-02-09 09:56:01] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.176:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK701bba97;rport From: "Client Care North" ;tag=as4ce7104a To: ;tag=9E7AD2E1-E1F94AA6 CSeq: 102 INVITE Call-ID: 130d41e27e4785013f4c42bc27084757@174.47.138.162 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.3.0439 Accept-Language: en Content-Type: application/sdp Content-Length: 205 v=0 o=- 1297174445 1297174445 IN IP4 192.168.125.176 s=Polycom IP Phone c=IN IP4 192.168.125.176 t=0 0 m=audio 2264 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [2011-02-09 09:56:01] VERBOSE[1280] logger.c: --- (12 headers 9 lines) --- [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Found RTP audio format 0 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Found RTP audio format 101 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Found audio description format PCMU for ID 0 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Found audio description format telephone-event for ID 101 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Peer audio RTP is at port 192.168.125.176:2264 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: list_route: hop: [2011-02-09 09:56:01] VERBOSE[1280] logger.c: set_destination: Parsing for address/port to send to [2011-02-09 09:56:01] VERBOSE[1280] logger.c: set_destination: set destination to 192.168.125.176, port 5060 [2011-02-09 09:56:01] VERBOSE[1280] logger.c: Transmitting (no NAT) to 192.168.125.176:5060: ACK sip:ctvsh_1346@192.168.125.176 SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK558c9161;rport From: "Client Care North" ;tag=as4ce7104a To: ;tag=9E7AD2E1-E1F94AA6 Contact: Call-ID: 130d41e27e4785013f4c42bc27084757@174.47.138.162 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [2011-02-09 09:56:02] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.20:5060 ---> SIP/2.0 200 OK To: ;tag=79070210bb8d1e01i0 From: "Client Care North" ;tag=as7ec73b33 Call-ID: 78062834706582427f49d6656dce2478@174.47.138.162 CSeq: 102 INVITE Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK49aa0332 Contact: CTVSH North Sipura Server: Linksys/SPA2102-5.2.10 Remote-Party-ID: CTVSH North Paging Sipura ;screen=yes;party=called Content-Length: 263 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 326731346 326731346 IN IP4 192.168.125.20 s=- c=IN IP4 192.168.125.20 t=0 0 m=audio 16394 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [2011-02-09 09:56:02] VERBOSE[1280] logger.c: --- (13 headers 13 lines) --- [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found RTP audio format 0 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found RTP audio format 100 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found RTP audio format 101 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found audio description format PCMU for ID 0 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found unknown media description format NSE for ID 100 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found audio description format telephone-event for ID 101 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Peer audio RTP is at port 192.168.125.20:16394 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: list_route: hop: [2011-02-09 09:56:02] VERBOSE[1280] logger.c: set_destination: Parsing for address/port to send to [2011-02-09 09:56:02] VERBOSE[1280] logger.c: set_destination: set destination to 192.168.125.20, port 5060 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Transmitting (no NAT) to 192.168.125.20:5060: ACK sip:ctvsh_9999@192.168.125.20:5060 SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK53648aa0;rport From: "Client Care North" ;tag=as7ec73b33 To: ;tag=79070210bb8d1e01i0 Contact: Call-ID: 78062834706582427f49d6656dce2478@174.47.138.162 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 -- a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [2011-02-09 09:56:02] VERBOSE[1280] logger.c: --- (12 headers 9 lines) --- [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found RTP audio format 0 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found RTP audio format 101 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found audio description format PCMU for ID 0 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Found audio description format telephone-event for ID 101 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Peer audio RTP is at port 192.168.125.146:2228 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: list_route: hop: [2011-02-09 09:56:02] VERBOSE[1280] logger.c: set_destination: Parsing for address/port to send to [2011-02-09 09:56:02] VERBOSE[1280] logger.c: set_destination: set destination to 192.168.125.146, port 5060 [2011-02-09 09:56:02] VERBOSE[1280] logger.c: Transmitting (no NAT) to 192.168.125.146:5060: ACK sip:ctvsh_1365@192.168.125.146 SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK648c2d16;rport From: "Client Care North" ;tag=as6caf1df2 To: ;tag=4E4FBDDD-29302AD0 Contact: Call-ID: 23c75d4a35e1201669f0297576c065d6@174.47.138.162 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [2011-02-09 09:56:02] VERBOSE[3296] logger.c: -- SIP/ctvsh_1365-000008e8 answered --- [2011-02-09 09:56:03] VERBOSE[3292] logger.c: -- Playing 'tmp/ctvsh_north-page-message-456011' (language 'en') [2011-02-09 09:56:07] VERBOSE[3292] logger.c: -- Playing 'tmp/ctvsh_north-page-message-456011' (language 'en') [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.175:5060: CANCEL sip:ctvsh_1345@192.168.125.175;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK21e3f48d;rport From: "Client Care North" ;tag=as5910996d To: Call-ID: 785f5a0e0ee454236e844e13301ca403@174.47.138.162 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Scheduling destruction of SIP dialog '785f5a0e0ee454236e844e13301ca403@174.47.138.162' in 6400 ms (Method: INVITE) [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Scheduling destruction of SIP dialog '23c75d4a35e1201669f0297576c065d6@174.47.138.162' in 6400 ms (Method: INVITE) [2011-02-09 09:56:11] VERBOSE[3293] logger.c: set_destination: Parsing for address/port to send to [2011-02-09 09:56:11] VERBOSE[3293] logger.c: set_destination: set destination to 192.168.125.146, port 5060 [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.146:5060: BYE sip:ctvsh_1365@192.168.125.146 SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK5109cf62;rport From: "Client Care North" ;tag=as6caf1df2 To: ;tag=4E4FBDDD-29302AD0 Call-ID: 23c75d4a35e1201669f0297576c065d6@174.47.138.162 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2011-02-09 09:56:11] VERBOSE[3297] logger.c: -- Hungup 'DAHDI/pseudo-2056476083' [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Scheduling destruction of SIP dialog '444209da526f6a966e87fd5c06a2e48d@174.47.138.162' in 6400 ms (Method: INVITE) [2011-02-09 09:56:11] VERBOSE[3293] logger.c: set_destination: Parsing for address/port to send to [2011-02-09 09:56:11] VERBOSE[3293] logger.c: set_destination: set destination to 192.168.125.180, port 5060 [2011-02-09 09:56:11] VERBOSE[3293] logger.c: Reliably Transmitting (no NAT) to 192.168.125.180:5060: BYE sip:ctvsh_1367@192.168.125.180 SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK53b89462;rport From: "Client Care North" ;tag=as02276705 To: ;tag=E00854E7-3259039C Call-ID: 444209da526f6a966e87fd5c06a2e48d@174.47.138.162 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [2011-02-09 09:56:11] VERBOSE[3293] logger.c: == Spawn extension (system-do-page-with-recording, s, 2) exited non-zero on 'Local/s@system-do-page-with-recording-c76a,2' [2011-02-09 09:56:11] NOTICE[3292] pbx_spool.c: Call completed to Local/s@system-do-page-with-recording/n [2011-02-09 09:56:11] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.175:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK21e3f48d;rport From: "Client Care North" ;tag=as5910996d To: CSeq: 102 CANCEL Call-ID: 785f5a0e0ee454236e844e13301ca403@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 [2011-02-09 09:56:11] VERBOSE[1280] logger.c: --- (11 headers 0 lines) --- [2011-02-09 09:56:11] VERBOSE[1280] logger.c: SIP Response message for INCOMING dialog NOTIFY arrived [2011-02-09 09:56:11] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.175:5060 ---> SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK21e3f48d;rport From: "Client Care North" ;tag=as5910996d To: ;tag=44258522-A6B87D25 CSeq: 102 INVITE Call-ID: 785f5a0e0ee454236e844e13301ca403@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 <-------------> [2011-02-09 09:56:11] VERBOSE[1280] logger.c: --- (10 headers 0 lines) --- [2011-02-09 09:56:11] VERBOSE[1280] logger.c: Transmitting (no NAT) to 192.168.125.175:5060: ACK sip:ctvsh_1345@192.168.125.175;transport=udp SIP/2.0 Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK21e3f48d;rport From: "Client Care North" ;tag=as5910996d To: ;tag=44258522-A6B87D25 Contact: Call-ID: 785f5a0e0ee454236e844e13301ca403@174.47.138.162 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [2011-02-09 09:56:11] VERBOSE[1280] logger.c: Really destroying SIP dialog '785f5a0e0ee454236e844e13301ca403@174.47.138.162' Method: INVITE [2011-02-09 09:56:11] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.146:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK5109cf62;rport From: "Client Care North" ;tag=as6caf1df2 To: ;tag=4E4FBDDD-29302AD0 CSeq: 103 BYE Call-ID: 23c75d4a35e1201669f0297576c065d6@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.1.3.0507 Accept-Language: en Content-Length: 0 <-------------> [2011-02-09 09:56:11] VERBOSE[1280] logger.c: --- (10 headers 0 lines) --- [2011-02-09 09:56:11] VERBOSE[1280] logger.c: <--- SIP read from 192.168.125.180:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 174.47.138.162:5060;branch=z9hG4bK53b89462;rport From: "Client Care North" ;tag=as02276705 To: ;tag=E00854E7-3259039C CSeq: 103 BYE Call-ID: 444209da526f6a966e87fd5c06a2e48d@174.47.138.162 Contact: User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.1.3.0507 Accept-Language: en Content-Length: 0 <-------------> [2011-02-09 09:56:11] VERBOSE[1280] logger.c: --- (10 headers 0 lines) --- [2011-02-09 09:56:11] VERBOSE[1280] logger.c: Really destroying SIP dialog '444209da526f6a966e87fd5c06a2e48d@174.47.138.162' Method: INVITE [2011-02-09 09:56:11] VERBOSE[1280] logger.c: Really destroying SIP dialog '23c75d4a35e1201669f0297576c065d6@174.47.138.162' Method: INVITE