Jul 16 00:03:53 VERBOSE[17984] logger.c: == Parsing '/etc/asterisk/manager.conf': Jul 16 00:03:53 VERBOSE[17984] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Jul 16 00:03:53 VERBOSE[17984] logger.c: == Manager 'FlashPanel' logged on from 192.168.44.13 Jul 16 00:05:13 VERBOSE[17972] logger.c: -- Accepting AUTHENTICATED call from 24.202.5.69: > requested format = ulaw, > requested prefs = (ulaw), > actual format = ulaw, > host prefs = (ulaw), > priority = mine Jul 16 00:05:13 VERBOSE[18007] logger.c: -- Executing Set("IAX2/321-4", "_ALERT_INFO=info=alert-autoanswer") in new stack Jul 16 00:05:13 VERBOSE[18007] logger.c: -- Executing Page("IAX2/321-4", "SIP/211&SIP/212") in new stack Jul 16 00:05:13 VERBOSE[18007] logger.c: -- Playing 'beep' (language 'fr') Jul 16 00:05:13 VERBOSE[18009] logger.c: We're at 192.168.44.13 port 10178 Jul 16 00:05:13 VERBOSE[18009] logger.c: Adding codec 0x4 (ulaw) to SDP Jul 16 00:05:13 VERBOSE[18009] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jul 16 00:05:13 VERBOSE[18009] logger.c: 14 headers, 10 lines Jul 16 00:05:13 VERBOSE[18009] logger.c: Reliably Transmitting (no NAT) to 192.168.44.72:5060: INVITE sip:211@192.168.44.72 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK4cd880e2;rport From: "David" ;tag=as22165c8c To: Contact: Call-ID: 66f237fc273245af0234ed5e70358bb9@192.168.44.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 16 Jul 2006 04:05:13 GMT Alert-Info: info=alert-autoanswer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 844 844 IN IP4 192.168.44.13 s=session c=IN IP4 192.168.44.13 t=0 0 m=audio 10178 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 16 00:05:13 VERBOSE[18010] logger.c: We're at 192.168.44.13 port 10108 Jul 16 00:05:13 VERBOSE[18010] logger.c: Adding codec 0x4 (ulaw) to SDP Jul 16 00:05:13 VERBOSE[18010] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Jul 16 00:05:13 VERBOSE[18010] logger.c: 14 headers, 10 lines Jul 16 00:05:13 VERBOSE[18010] logger.c: Reliably Transmitting (no NAT) to 192.168.44.53:5060: INVITE sip:212@192.168.44.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK10ac8867;rport From: "David" ;tag=as59af9f74 To: Contact: Call-ID: 4f81a35b2c4330e644fcdb282bcd45ce@192.168.44.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 16 Jul 2006 04:05:13 GMT Alert-Info: info=alert-autoanswer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 o=root 844 844 IN IP4 192.168.44.13 s=session c=IN IP4 192.168.44.13 t=0 0 m=audio 10108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.72:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK4cd880e2;rport From: "David" ;tag=as22165c8c To: ;tag=355A048C-88E042A1 CSeq: 102 INVITE Call-ID: 66f237fc273245af0234ed5e70358bb9@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (9 headers 0 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (9 headers 0 lines)--- Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.53:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK10ac8867;rport From: "David" ;tag=as59af9f74 To: ;tag=B634925D-2C0A71CA CSeq: 102 INVITE Call-ID: 4f81a35b2c4330e644fcdb282bcd45ce@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (9 headers 0 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (9 headers 0 lines)--- Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.72:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK4cd880e2;rport From: "David" ;tag=as22165c8c To: ;tag=355A048C-88E042A1 CSeq: 102 INVITE Call-ID: 66f237fc273245af0234ed5e70358bb9@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (10 headers 0 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (10 headers 0 lines)--- Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.53:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK10ac8867;rport From: "David" ;tag=as59af9f74 To: ;tag=B634925D-2C0A71CA CSeq: 102 INVITE Call-ID: 4f81a35b2c4330e644fcdb282bcd45ce@192.168.44.13 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (10 headers 0 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (10 headers 0 lines)--- Jul 16 00:05:13 VERBOSE[18007] logger.c: == Spawn extension (internal, 11112, 2) exited non-zero on 'IAX2/321-4' Jul 16 00:05:13 VERBOSE[18007] logger.c: -- Hungup 'IAX2/321-4' Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.53:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK10ac8867;rport From: "David" ;tag=as59af9f74 To: ;tag=B634925D-2C0A71CA CSeq: 102 INVITE Call-ID: 4f81a35b2c4330e644fcdb282bcd45ce@192.168.44.13 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1153022781 1153022781 IN IP4 192.168.44.53 s=Polycom IP Phone c=IN IP4 192.168.44.53 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (11 headers 8 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (11 headers 8 lines)--- Jul 16 00:05:13 VERBOSE[17971] logger.c: Found RTP audio format 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: Found RTP audio format 101 Jul 16 00:05:13 VERBOSE[17971] logger.c: Peer audio RTP is at port 192.168.44.53:2222 Jul 16 00:05:13 VERBOSE[17971] logger.c: Found description format PCMU Jul 16 00:05:13 VERBOSE[17971] logger.c: Found description format telephone-event Jul 16 00:05:13 VERBOSE[17971] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jul 16 00:05:13 VERBOSE[17971] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 16 00:05:13 VERBOSE[17971] logger.c: list_route: hop: Jul 16 00:05:13 VERBOSE[17971] logger.c: set_destination: Parsing for address/port to send to Jul 16 00:05:13 VERBOSE[17971] logger.c: set_destination: set destination to 192.168.44.53, port 5060 Jul 16 00:05:13 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.53:5060: ACK sip:212@192.168.44.53 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK317cd85b;rport From: "David" ;tag=as59af9f74 To: ;tag=B634925D-2C0A71CA Contact: Call-ID: 4f81a35b2c4330e644fcdb282bcd45ce@192.168.44.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 16 00:05:13 VERBOSE[18014] logger.c: -- Launching MeetMe(1876180073d|mqxdw) on SIP/212-c06d Jul 16 00:05:13 VERBOSE[18014] logger.c: -- Created MeetMe conference 1023 for conference '1876180073d' Jul 16 00:05:13 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.72:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK4cd880e2;rport From: "David" ;tag=as22165c8c To: ;tag=355A048C-88E042A1 CSeq: 102 INVITE Call-ID: 66f237fc273245af0234ed5e70358bb9@192.168.44.13 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1153022770 1153022770 IN IP4 192.168.44.72 s=Polycom IP Phone c=IN IP4 192.168.44.72 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (11 headers 8 lines)Jul 16 00:05:13 VERBOSE[17971] logger.c: --- (11 headers 8 lines)--- Jul 16 00:05:13 VERBOSE[17971] logger.c: Found RTP audio format 0 Jul 16 00:05:13 VERBOSE[17971] logger.c: Found RTP audio format 101 Jul 16 00:05:13 VERBOSE[17971] logger.c: Peer audio RTP is at port 192.168.44.72:2222 Jul 16 00:05:13 VERBOSE[17971] logger.c: Found description format PCMU Jul 16 00:05:13 VERBOSE[17971] logger.c: Found description format telephone-event Jul 16 00:05:13 VERBOSE[17971] logger.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jul 16 00:05:13 VERBOSE[17971] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 16 00:05:13 VERBOSE[17971] logger.c: list_route: hop: Jul 16 00:05:13 VERBOSE[17971] logger.c: set_destination: Parsing for address/port to send to Jul 16 00:05:13 VERBOSE[17971] logger.c: set_destination: set destination to 192.168.44.72, port 5060 Jul 16 00:05:13 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.72:5060: ACK sip:211@192.168.44.72 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK0a3814be;rport From: "David" ;tag=as22165c8c To: ;tag=355A048C-88E042A1 Contact: Call-ID: 66f237fc273245af0234ed5e70358bb9@192.168.44.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 16 00:05:13 VERBOSE[18017] logger.c: -- Launching MeetMe(1876180073d|mqxdw) on SIP/211-76c3 Jul 16 00:05:21 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.69:5060: REGISTER sip:192.168.44.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK93b784c64 Max-Forwards: 70 Content-Length: 0 To: Jeffrey From: Jeffrey ;tag=a0f01a172c04757 Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796978 REGISTER Contact: Jeffrey Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Authorization:Digest response="1be8b7788e125b6b1f2b3cc5f4a6caf6",username="233",realm="asterisk",nonce="3462872d",uri="sip:192.168.44.13:5060" User-Agent: Aastra 480i/1.3.1.1095 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Jul 16 00:05:21 VERBOSE[17971] logger.c: --- (13 headers 0 lines)Jul 16 00:05:21 VERBOSE[17971] logger.c: --- (13 headers 0 lines)--- Jul 16 00:05:21 VERBOSE[17971] logger.c: Using latest REGISTER request as basis request Jul 16 00:05:21 VERBOSE[17971] logger.c: Sending to 192.168.44.69 : 5060 (non-NAT) Jul 16 00:05:21 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.69:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK93b784c64;received=192.168.44.69 From: Jeffrey ;tag=a0f01a172c04757 To: Jeffrey Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796978 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 16 00:05:21 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.69:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK93b784c64;received=192.168.44.69 From: Jeffrey ;tag=a0f01a172c04757 To: Jeffrey ;tag=as3a2e4103 Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796978 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="5dc9099a" Content-Length: 0 --- Jul 16 00:05:21 VERBOSE[17971] logger.c: Scheduling destruction of call '2594e2daa988fb6996e6c99b149043cb@192.168.44.69' in 15000 ms Jul 16 00:05:21 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.69:5060: REGISTER sip:192.168.44.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK678378904 Max-Forwards: 70 Content-Length: 0 To: Jeffrey From: Jeffrey ;tag=a0f01a172c04757 Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796979 REGISTER Contact: Jeffrey Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Authorization:Digest response="97146d7fb511bbfd6e44c549a67b1211",username="233",realm="asterisk",nonce="5dc9099a",uri="sip:192.168.44.13:5060" User-Agent: Aastra 480i/1.3.1.1095 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Jul 16 00:05:21 VERBOSE[17971] logger.c: --- (13 headers 0 lines)Jul 16 00:05:21 VERBOSE[17971] logger.c: --- (13 headers 0 lines)--- Jul 16 00:05:21 VERBOSE[17971] logger.c: Using latest REGISTER request as basis request Jul 16 00:05:21 VERBOSE[17971] logger.c: Sending to 192.168.44.69 : 5060 (non-NAT) Jul 16 00:05:21 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.69:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK678378904;received=192.168.44.69 From: Jeffrey ;tag=a0f01a172c04757 To: Jeffrey Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796979 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 16 00:05:21 VERBOSE[17971] logger.c: -- Saved useragent "Aastra 480i/1.3.1.1095 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26" for peer 233 Jul 16 00:05:21 VERBOSE[17971] logger.c: Transmitting (no NAT) to 192.168.44.69:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.44.69;branch=z9hG4bK678378904;received=192.168.44.69 From: Jeffrey ;tag=a0f01a172c04757 To: Jeffrey ;tag=as3a2e4103 Call-ID: 2594e2daa988fb6996e6c99b149043cb@192.168.44.69 CSeq: 424796979 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: ;expires=120 Date: Sun, 16 Jul 2006 04:05:21 GMT Content-Length: 0 --- Jul 16 00:05:21 VERBOSE[17971] logger.c: Scheduling destruction of call '2594e2daa988fb6996e6c99b149043cb@192.168.44.69' in 15000 ms Jul 16 00:05:24 VERBOSE[17971] logger.c: 12 headers, 3 lines Jul 16 00:05:24 VERBOSE[17971] logger.c: Reliably Transmitting (no NAT) to 192.168.44.69:5060: NOTIFY sip:233@192.168.44.69 SIP/2.0 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK068e563d;rport From: "asterisk" ;tag=as2e1264c5 To: Contact: Call-ID: 6d3ba7273b6dfe6206a4eb353e7e3b17@192.168.44.13 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: yes Message-Account: sip:asterisk@192.168.44.13 Voice-Message: 2/2 (0/0) --- Jul 16 00:05:24 VERBOSE[17971] logger.c: Scheduling destruction of call '6d3ba7273b6dfe6206a4eb353e7e3b17@192.168.44.13' in 15000 ms Jul 16 00:05:24 VERBOSE[17971] logger.c: <-- SIP read from 192.168.44.69:5060: SIP/2.0 200 OK Call-ID: 6d3ba7273b6dfe6206a4eb353e7e3b17@192.168.44.13 CSeq: 102 NOTIFY From: "asterisk" ;tag=as2e1264c5 To: ;tag=b959b7869ae0b78 Via: SIP/2.0/UDP 192.168.44.13:5060;branch=z9hG4bK068e563d;rport Content-Length: 0 Contact: Supported: replaces User-Agent: Aastra 480i/1.3.1.1095 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 Jul 16 00:05:24 VERBOSE[17971] logger.c: --- (10 headers 0 lines)Jul 16 00:05:24 VERBOSE[17971] logger.c: --- (10 headers 0 lines)--- Jul 16 00:05:24 VERBOSE[17971] logger.c: Destroying call '6d3ba7273b6dfe6206a4eb353e7e3b17@192.168.44.13' Jul 16 00:05:47 VERBOSE[17979] logger.c: -- Remote UNIX connection disconnected