burnabyasterisk*CLI> SIP Debugging re-enabled burnabyasterisk*CLI> [2011-08-26 12:45:22] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK82a8fcf35B9E9EE2 From: "Boardroom" ;tag=2F71B4D-2BCF2DAC To: CSeq: 1 SUBSCRIBE Call-ID: fa453339-9a0dd3b8-d3fd041f@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 12:45:22] --- (15 headers 0 lines) --- burnabyasterisk*CLI> [2011-08-26 12:45:22] DEBUG[21911]: acl.c:725 ast_ouraddrfor: For destination '172.30.0.189', our source address is '172.30.0.10'. [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:3479 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.30.0.10:5060 burnabyasterisk*CLI> [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for fa453339-9a0dd3b8-d3fd041f@172.30.0.189 - SUBSCRIBE (No RTP) [2011-08-26 12:45:22] Creating new subscription burnabyasterisk*CLI> [2011-08-26 12:45:22] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:13743 build_route: build_route: Contact hop: burnabyasterisk*CLI> [2011-08-26 12:45:22] list_route: hop: burnabyasterisk*CLI> [2011-08-26 12:45:22] Found peer '4109' for '4109' from 172.30.0.189:5060 burnabyasterisk*CLI> [2011-08-26 12:45:22] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK82a8fcf35B9E9EE2;received=172.30.0.189 From: "Boardroom" ;tag=2F71B4D-2BCF2DAC To: ;tag=as06ac1c25 Call-ID: fa453339-9a0dd3b8-d3fd041f@172.30.0.189 CSeq: 1 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH S burnabyasterisk*CLI> upported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c778867" Content-Length: 0 <------------> [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 172.30.0.189:5060 [2011-08-26 12:45:22] Scheduling destruction of SIP dialog 'fa453339-9a0dd3b8-d3fd041f@172.30.0.189' in 6400 ms (Method: SUBSCRIBE) burnabyasterisk*CLI> [2011-08-26 12:45:22] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7d4a22e916C55A5 From: "Boardroom" ;tag=2F71B4D-2BCF2DAC To: CSeq: 2 SUBSCRIBE Call-ID: fa453339-9a0dd3b8-d3fd041f@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Authorization: Digest username="4109", realm="asterisk", nonce="3c778867", uri="sip:4109@burnabyasterisk.customer.com:5060", response="dbded3f6d6d555d8f0f53b241174a24f", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 12:45:22] --- (16 headers 0 lines) --- [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:23931 handle_request_subscribe: Got a new subscription fa453339-9a0dd3b8-d3fd041f@172.30.0.189 (possibly with auth) or retransmission [2011-08-26 12:45:22] Creating new subscription [2011-08-26 12:45:22] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:13680 build_route: build_route: Retaining previous route: [2011-08-26 12:45:22] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 12:45:22] Looking for 4109 in default-local (domain burnabyasterisk.customer.com:5060) [2011-08-26 12:45:22] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7d4a22e916C55A5;received=172.30.0.189 From: "Boardroom" ;tag=2F71B4D-2BCF2DAC To: ;tag=as06ac1c25 Call-ID: fa453339-9a0dd3b8-d3fd041f@172.30.0.189 CSeq: 2 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 489' onto UDP socket destined for 172.30.0.189:5060 [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:24180 handle_request_subscribe: Received SIP subscribe for unknown event package: missed-call-summary [2011-08-26 12:45:22] DEBUG[21911]: chan_sip.c:5898 sip_destroy: Destroying SIP dialog fa453339-9a0dd3b8-d3fd041f@172.30.0.189 [2011-08-26 12:45:22] Really destroying SIP dialog 'fa453339-9a0dd3b8-d3fd041f@172.30.0.189' Method: SUBSCRIBE burnabyasterisk*CLI> [2011-08-26 12:45:24] <--- SIP read from UDP:172.30.0.188:5060 ---> INVITE sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKb5afce90287EAA85 From: "4149" ;tag=A33E758E-7D4FEB53 To: CSeq: 1 INVITE Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1314387920 1314387920 IN IP4 172.30.0.188 s=Polycom IP Phone c=IN IP4 172.30.0.188 t=0 0 a=sendrecv m=audio 2224 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 <-------------> [2011-08-26 12:45:24] --- (15 headers 13 lines) --- [2011-08-26 12:45:24] DEBUG[21911]: acl.c:725 ast_ouraddrfor: For destination '172.30.0.188', our source address is '172.30.0.10'. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:3479 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.30.0.10:5060 [2011-08-26 12:45:24] == Using UDPTL TOS bits 184 [2011-08-26 12:45:24] == Using UDPTL CoS mark 5 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to Off [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for ab78d872-b933cf97-74af534@172.30.0.188 - INVITE (No RTP) [2011-08-26 12:45:24] DEBUG[21911]: sip/reqresp_parser.c:1613 parse_sip_options: Begin: parsing SIP "Supported: 100rel,replaces" [2011-08-26 12:45:24] DEBUG[21911]: sip/reqresp_parser.c:1629 parse_sip_options: Found SIP option: -100rel- [2011-08-26 12:45:24] DEBUG[21911]: sip/reqresp_parser.c:1637 parse_sip_options: Matched SIP option: 100rel [2011-08-26 12:45:24] DEBUG[21911]: sip/reqresp_parser.c:1629 parse_sip_options: Found SIP option: -replaces- [2011-08-26 12:45:24] DEBUG[21911]: sip/reqresp_parser.c:1637 parse_sip_options: Matched SIP option: replaces [2011-08-26 12:45:24] Sending to 172.30.0.188:5060 (no NAT) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:21875 handle_request_invite: Initializing initreq for method INVITE - callid ab78d872-b933cf97-74af534@172.30.0.188 [2011-08-26 12:45:24] Using INVITE request as basis request - ab78d872-b933cf97-74af534@172.30.0.188 [2011-08-26 12:45:24] Found peer '4149' for '4149' from 172.30.0.188:5060 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to Off [2011-08-26 12:45:24] <--- Reliably Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKb5afce90287EAA85;received=172.30.0.188 From: "4149" ;tag=A33E758E-7D4FEB53 To: ;tag=as5313efea Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 CSeq: 1 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ed0003f" Content-Length: 0 <------------> [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 172.30.0.188:5060 [2011-08-26 12:45:24] Scheduling destruction of SIP dialog 'ab78d872-b933cf97-74af534@172.30.0.188' in 6400 ms (Method: INVITE) burnabyasterisk*CLI> [2011-08-26 12:45:24] <--- SIP read from UDP:172.30.0.188:5060 ---> ACK sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKb5afce90287EAA85 From: "4149" ;tag=A33E758E-7D4FEB53 To: ;tag=as5313efea CSeq: 1 ACK Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-08-26 12:45:24] --- (12 headers 0 lines) --- [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4011 __sip_ack: Stopping retransmission on 'ab78d872-b933cf97-74af534@172.30.0.188' of Response 1: Match Found burnabyasterisk*CLI> [2011-08-26 12:45:24] <--- SIP read from UDP:172.30.0.188:5060 ---> INVITE sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK1a6fec895E3266D6 From: "4149" ;tag=A33E758E-7D4FEB53 To: CSeq: 2 INVITE Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="4149", realm="asterisk", nonce="0ed0003f", uri="sip:*800@burnabyasterisk.customer.com:5060;user=phone", response="be5bc0489c5e712255d4804cff4c2824", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1314387920 1314387920 IN IP4 172.30.0.188 s=Polycom IP Phone c=IN IP4 172.30.0.188 t=0 0 a=sendrecv m=audio 2224 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 <-------------> burnabyasterisk*CLI> [2011-08-26 12:45:24] --- (16 headers 13 lines) --- [2011-08-26 12:45:24] Sending to 172.30.0.188:5060 (no NAT) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:21875 handle_request_invite: Initializing initreq for method INVITE - callid ab78d872-b933cf97-74af534@172.30.0.188 [2011-08-26 12:45:24] Using INVITE request as basis request - ab78d872-b933cf97-74af534@172.30.0.188 [2011-08-26 12:45:24] Found peer '4149' for '4149' from 172.30.0.188:5060 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to Off [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:345 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x8c197e0' [2011-08-26 12:45:24] DEBUG[21911]: res_rtp_asterisk.c:483 ast_rtp_new: Allocated port 19330 for RTP instance '0x8c197e0' [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:354 ast_rtp_instance_new: RTP instance '0x8c197e0' is setup and ready to go burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21911]: res_rtp_asterisk.c:2394 ast_rtp_prop_set: Setup RTCP on RTP instance '0x8c197e0' [2011-08-26 12:45:24] == Using SIP RTP TOS bits 184 [2011-08-26 12:45:24] == Using SIP RTP CoS mark 5 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4934 do_setnat: Setting NAT on RTP to Off [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4942 do_setnat: Setting NAT on UDPTL to Off [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP o=- 1314387920 1314387920 IN IP4 172.30.0.188... UNSUPPORTED. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP c=IN IP4 172.30.0.188... OK. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8626 process_sdp: Processing session-level SDP a=sendrecv... OK. [2011-08-26 12:45:24] Found RTP audio format 9 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0xb791a760 [2011-08-26 12:45:24] Found RTP audio format 0 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0xb791a760 [2011-08-26 12:45:24] Found RTP audio format 8 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0xb791a760 [2011-08-26 12:45:24] Found RTP audio format 18 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on 0xb791a760 [2011-08-26 12:45:24] Found RTP audio format 101 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:536 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0xb791a760 [2011-08-26 12:45:24] Found audio description format G722 for ID 9 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [2011-08-26 12:45:24] Found audio description format PCMU for ID 0 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2011-08-26 12:45:24] Found audio description format PCMA for ID 8 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2011-08-26 12:45:24] Found audio description format G729 for ID 18 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED. [2011-08-26 12:45:24] Found audio description format telephone-event for ID 101 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:8813 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0xb791a760 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0xb791a760 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 9 on 0xb791a760 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0xb791a760 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:639 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0xb791a760 [2011-08-26 12:45:24] Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1004 (ulaw|g722) [2011-08-26 12:45:24] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-08-26 12:45:24] DEBUG[21911]: res_rtp_asterisk.c:2415 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x8c197e0' [2011-08-26 12:45:24] Peer audio RTP is at port 172.30.0.188:2224 [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0xb791a760 to 0x8c1998c [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0xb791a760 to 0x8c1998c [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0xb791a760 to 0x8c1998c [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 18 from 0xb791a760 to 0x8c1998c [2011-08-26 12:45:24] DEBUG[21911]: rtp_engine.c:517 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0xb791a760 to 0x8c1998c [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:9035 process_sdp: Peer doesn't provide T.38 UDPTL [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:9045 process_sdp: We're settling with these formats: 0x1004 (ulaw|g722) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:22023 handle_request_invite: Checking SIP call limits for device 4149 [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:5750 update_call_counter: Updating call counter for incoming call [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:5855 update_call_counter: Call from peer '4149' is 2 out of 8 [2011-08-26 12:45:24] DEBUG[21902]: chan_sip.c:25675 sip_devicestate: Checking device state for peer 4149 [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:458 do_state_change: Changing state for SIP/4149 - state 2 (In use) [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:438 devstate_event: device 'SIP/4149' state '2' [2011-08-26 12:45:24] Looking for *800 in default-super (domain burnabyasterisk.customer.com:5060) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:6845 sip_new: *** Our native formats are 0x1000 (g722) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:6846 sip_new: *** Joint capabilities are 0x1004 (ulaw|g722) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:6847 sip_new: *** Our capabilities are 0x1004 (ulaw|g722) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:6848 sip_new: *** AST_CODEC_CHOOSE formats are 0x1000 (g722) [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:6878 sip_new: This channel will not be able to handle video. [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:13743 build_route: build_route: Contact hop: [2011-08-26 12:45:24] list_route: hop: [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:22315 handle_request_invite: SIP/4149-00000005: New call is still down.... Trying... [2011-08-26 12:45:24] <--- Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK1a6fec895E3266D6;received=172.30.0.188 From: "4149" ;tag=A33E758E-7D4FEB53 To: Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 172.30.0.188:5060 [2011-08-26 12:45:24] DEBUG[21902]: chan_sip.c:25675 sip_devicestate: Checking device state for peer 4149 [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:458 do_state_change: Changing state for SIP/4149 - state 2 (In use) [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:438 devstate_event: device 'SIP/4149' state '2' [2011-08-26 12:45:24] DEBUG[21840]: cel_radius.c:191 radius_log: Unable to create RADIUS record. CEL not recorded! [2011-08-26 12:45:24] DEBUG[21937]: app_queue.c:1492 handle_statechange: Device 'SIP/4149' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-08-26 12:45:24] DEBUG[30010]: pbx.c:3929 pbx_substitute_variables_helper_full: Function result is '4149' [2011-08-26 12:45:24] DEBUG[30010]: pbx.c:4101 pbx_extension_helper: Launching 'Set' [2011-08-26 12:45:24] -- Executing [*800@default-super:1] Set("SIP/4149-00000005", "GROUP(OUTGOING)=4149") in new stack [2011-08-26 12:45:24] DEBUG[21840]: res_odbc.c:1035 odbc_release_obj2: odbc_release_obj2(0x875fb08) called (obj->txf = (nil)) [2011-08-26 12:45:24] DEBUG[21937]: app_queue.c:1492 handle_statechange: Device 'SIP/4149' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-08-26 12:45:24] DEBUG[30010]: pbx.c:4101 pbx_extension_helper: Launching 'Answer' [2011-08-26 12:45:24] -- Executing [*800@default-super:2] Answer("SIP/4149-00000005", "") in new stack burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21902]: chan_sip.c:25675 sip_devicestate: Checking device state for peer 4149 [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:458 do_state_change: Changing state for SIP/4149 - state 2 (In use) [2011-08-26 12:45:24] DEBUG[21902]: devicestate.c:438 devstate_event: device 'SIP/4149' state '2' [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:6324 sip_answer: SIP answering channel: SIP/4149-00000005 burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21937]: app_queue.c:1492 handle_statechange: Device 'SIP/4149' changed to state '2' (In use) but we don't care because they're not a member of any queue. [2011-08-26 12:45:24] DEBUG[30010]: res_rtp_asterisk.c:749 ast_rtp_update_source: Setting the marker bit due to a source update burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:11360 transmit_response_with_sdp: Setting framing from config on incoming call burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:11006 add_sdp: ** Our capability: 0x1004 (ulaw|g722) Video flag: True Text flag: True burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:11007 add_sdp: ** Our prefcodec: 0x0 (nothing) burnabyasterisk*CLI> [2011-08-26 12:45:24] Audio is at 5060 burnabyasterisk*CLI> [2011-08-26 12:45:24] Adding codec 0x1000 (g722) to SDP burnabyasterisk*CLI> [2011-08-26 12:45:24] Adding codec 0x4 (ulaw) to SDP burnabyasterisk*CLI> [2011-08-26 12:45:24] Adding non-codec 0x1 (telephone-event) to SDP burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:11116 add_sdp: -- Done with adding codecs to SDP burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:11255 add_sdp: Done building SDP. Settling with this capability: 0x1004 (ulaw|g722) burnabyasterisk*CLI> [2011-08-26 12:45:24] <--- Reliably Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK1a6fec895E3266D6;received=172.30.0.188 From: "4149" ;tag=A33E758E-7D4FEB53 To: ;tag=as6b87fd00 Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 1405442783 1405442783 IN IP4 172.30.0.10 s=Asterisk PBX 1.8.6.0 c=IN IP4 172.30.0.10 t=0 0 m=audio 19330 RTP/AVP 9 0 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 172.30.0.188:5060 burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21840]: cel_radius.c:191 radius_log: Unable to create RADIUS record. CEL not recorded! burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21840]: res_odbc.c:1035 odbc_release_obj2: odbc_release_obj2(0x875fb08) called (obj->txf = (nil)) burnabyasterisk*CLI> [2011-08-26 12:45:24] <--- SIP read from UDP:172.30.0.188:5060 ---> ACK sip:*800@172.30.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK170b60d38A4B0BA From: "4149" ;tag=A33E758E-7D4FEB53 To: ;tag=as6b87fd00 CSeq: 2 ACK Call-ID: ab78d872-b933cf97-74af534@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-08-26 12:45:24] --- (12 headers 0 lines) --- burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[21911]: chan_sip.c:4011 __sip_ack: Stopping retransmission on 'ab78d872-b933cf97-74af534@172.30.0.188' of Response 2: Match Found burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: pbx.c:4101 pbx_extension_helper: Launching 'Set' burnabyasterisk*CLI> [2011-08-26 12:45:24] -- Executing [*800@default-super:3] Set("SIP/4149-00000005", "CHANNEL(musicclass)=default_default") in new stack burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: pbx.c:4101 pbx_extension_helper: Launching 'MeetMe' burnabyasterisk*CLI> [2011-08-26 12:45:24] -- Executing [*800@default-super:4] MeetMe("SIP/4149-00000005", "1,TMi") in new stack burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: app_meetme.c:3994 find_conf: The requested confno is '1'? burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: app_meetme.c:3997 find_conf: Does conf 1 match 1? burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: res_rtp_asterisk.c:1263 ast_rtp_write: Ooh, format changed from unknown to g722 burnabyasterisk*CLI> [2011-08-26 12:45:24] DEBUG[30010]: res_rtp_asterisk.c:1294 ast_rtp_write: Created smoother: format: g722 ms: 20 len: 160 [2011-08-26 12:45:24] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:24] -- Playing 'conf-getpin.g722' (language 'en') burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 [2011-08-26 12:45:25] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:25] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 51 (3), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 51 (3), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 56 (8), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:25] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 56 (8), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 51 (3), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 51 (3), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 56 (8), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 56 (8), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000008 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:26] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 54 (6), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 00000006 (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 35 (#), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 35 (#), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:27] -- Recording burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: app.c:767 __ast_play_and_record: play_and_record: vm-rec-name, /var/spool/asterisk/meetme/meetme-username-1-2, 'sln' burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1085 ast_rtp_raw_write: Difference is 12392, ms is 1569 burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-08-26 12:45:27] -- Playing 'vm-rec-name.g722' (language 'en') burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:27] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:29] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK5e796350EA9BA9F7 From: "Boardroom" ;tag=534E83FA-BCFA3291 To: CSeq: 1 SUBSCRIBE Call-ID: 1b77446-a3e679fd-3f2cacdc@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 12:45:29] --- (15 headers 0 lines) --- burnabyasterisk*CLI> [2011-08-26 12:45:29] DEBUG[21911]: acl.c:725 ast_ouraddrfor: For destination '172.30.0.189', our source address is '172.30.0.10'. [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:3479 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.30.0.10:5060 burnabyasterisk*CLI> [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for 1b77446-a3e679fd-3f2cacdc@172.30.0.189 - SUBSCRIBE (No RTP) [2011-08-26 12:45:29] Creating new subscription [2011-08-26 12:45:29] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:13743 build_route: build_route: Contact hop: [2011-08-26 12:45:29] list_route: hop: [2011-08-26 12:45:29] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 12:45:29] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK5e796350EA9BA9F7;received=172.30.0.189 From: "Boardroom" ;tag=534E83FA-BCFA3291 To: ;tag=as40656e51 Call-ID: 1b77446-a3e679fd-3f2cacdc@172.30.0.189 CSeq: 1 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6e93a742" Content-Length: 0 <------------> [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 172.30.0.189:5060 [2011-08-26 12:45:29] Scheduling destruction of SIP dialog '1b77446-a3e679fd-3f2cacdc@172.30.0.189' in 6400 ms (Method: SUBSCRIBE) burnabyasterisk*CLI> [2011-08-26 12:45:29] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK8d85a8a36674A312 From: "Boardroom" ;tag=534E83FA-BCFA3291 To: CSeq: 2 SUBSCRIBE Call-ID: 1b77446-a3e679fd-3f2cacdc@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Authorization: Digest username="4109", realm="asterisk", nonce="6e93a742", uri="sip:4109@burnabyasterisk.customer.com:5060", response="c182fabae0fa4283fbb9ec037875786d", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 12:45:29] --- (16 headers 0 lines) --- [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:23931 handle_request_subscribe: Got a new subscription 1b77446-a3e679fd-3f2cacdc@172.30.0.189 (possibly with auth) or retransmission [2011-08-26 12:45:29] Creating new subscription [2011-08-26 12:45:29] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:13680 build_route: build_route: Retaining previous route: [2011-08-26 12:45:29] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 12:45:29] Looking for 4109 in default-local (domain burnabyasterisk.customer.com:5060) [2011-08-26 12:45:29] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK8d85a8a36674A312;received=172.30.0.189 From: "Boardroom" ;tag=534E83FA-BCFA3291 To: ;tag=as40656e51 Call-ID: 1b77446-a3e679fd-3f2cacdc@172.30.0.189 CSeq: 2 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'SIP/2.0 489' onto UDP socket destined for 172.30.0.189:5060 [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:24180 handle_request_subscribe: Received SIP subscribe for unknown event package: missed-call-summary [2011-08-26 12:45:29] DEBUG[21911]: chan_sip.c:5898 sip_destroy: Destroying SIP dialog 1b77446-a3e679fd-3f2cacdc@172.30.0.189 [2011-08-26 12:45:29] Really destroying SIP dialog '1b77446-a3e679fd-3f2cacdc@172.30.0.189' Method: SUBSCRIBE burnabyasterisk*CLI> [2011-08-26 12:45:29] DEBUG[30010]: res_rtp_asterisk.c:1697 ast_rtcp_read: Got RTCP report of 84 bytes burnabyasterisk*CLI> [2011-08-26 12:45:30] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (61 requested / 61 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:30] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:30] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:30] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:30] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-08-26 12:45:30] -- Playing 'beep.g722' (language 'en') burnabyasterisk*CLI> [2011-08-26 12:45:31] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (182 requested / 182 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:31] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:31] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:31] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:31] DEBUG[30010]: app.c:791 __ast_play_and_record: Recording Formats: sfmts=sln burnabyasterisk*CLI> [2011-08-26 12:45:31] -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-1-2 format: sln, 0x8c1d838 burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 35 (#), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1402 create_dtmf_frame: Sending dtmf: 35 (#), at 172.30.0.188:2224 burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:32] -- User ended message by pressing # burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1085 ast_rtp_raw_write: Difference is 10736, ms is 1362 burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-08-26 12:45:32] -- Playing 'auth-thankyou.g722' (language 'en') burnabyasterisk*CLI> [2011-08-26 12:45:32] DEBUG[30010]: res_rtp_asterisk.c:1446 process_dtmf_rfc2833: - RTP 2833 Event: 0000000b (len = 4) burnabyasterisk*CLI> [2011-08-26 12:45:33] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:33] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:33] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-08-26 12:45:33] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second burnabyasterisk*CLI> [2011-08-26 12:45:33] DEBUG[30010]: channel.c:3480 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-08-26 12:45:33] -- Playing 'vm-review.g722' (language 'en') burnabyasterisk*CLI> [2011-08-26 12:45:34] DEBUG[21911]: chan_sip.c:7515 sip_alloc: Allocating new SIP dialog for 663e811036f840a97ae6776d21c4ba3e@192.168.200.3:0 - OPTIONS (No RTP) [2011-08-26 12:45:34] DEBUG[21911]: acl.c:725 ast_ouraddrfor: For destination '172.30.0.188', our source address is '172.30.0.10'. [2011-08-26 12:45:34] DEBUG[21911]: chan_sip.c:3479 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 172.30.0.10:5060 [2011-08-26 12:45:34] DEBUG[21911]: chan_sip.c:3054 initialize_initreq: Initializing initreq for method OPTIONS - callid 2c6e960821375cfb128e348f054ec969@172.30.0.10:5060 [2011-08-26 12:45:34] Reliably Transmitting (no NAT) to 172.30.0.188:5060: OPTIONS sip:4149@172.30.0.188:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK7685d5ec Max-Forwards: 70 From: "asterisk" ;tag=as00b087d0 To: Contact: Call-ID: 2c6e960821375cfb128e348f054ec969@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 19:45:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 12:45:34] DEBUG[21911]: chan_sip.c:3325 __sip_xmit: Trying to put 'OPTIONS sip' onto UDP socket destined for 172.30.0.188:5060 burnabyasterisk*CLI> Disconnected from Asterisk server [Aug 26 12:45:35] Executing last minute cleanups Asterisk ending (0). ]0;root@burnabyasterisk:~[root@burnabyasterisk ~]#