Dec 25 13:11:38 VERBOSE[786] logger.c: Destroying call 'a5ac6de9-feb81d5b-553806f0@192.168.254.252' Dec 25 13:11:41 DEBUG[786] chan_sip.c: Auto destroying call '13b841321b853b17725522b9445dc81b@192.168.254.250' Dec 25 13:11:41 VERBOSE[786] logger.c: Destroying call '13b841321b853b17725522b9445dc81b@192.168.254.250' Dec 25 13:11:41 DEBUG[786] chan_sip.c: Auto destroying call '6b4c68ed04787ddc2615ba2b36186ed6@192.168.254.250' Dec 25 13:11:41 VERBOSE[786] logger.c: Destroying call '6b4c68ed04787ddc2615ba2b36186ed6@192.168.254.250' Dec 25 13:11:41 DEBUG[786] chan_sip.c: Auto destroying call '797982a73a30dbb55f99ce4148f33e5e@192.168.254.250' Dec 25 13:11:41 VERBOSE[786] logger.c: Destroying call '797982a73a30dbb55f99ce4148f33e5e@192.168.254.250' Dec 25 13:11:48 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: REGISTER sip:cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-233d805fde66af54-1--d87543-;rport Max-Forwards: 70 Contact: To: "Kai-Uwe Jensen" From: "Kai-Uwe Jensen";tag=1373ec14 Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 0: REGISTER sip:cojensen.net SIP/2.0 (33) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-233d805fde66af54-1--d87543-;rport (92) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 2: Max-Forwards: 70 (16) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 3: Contact: (42) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 4: To: "Kai-Uwe Jensen" (46) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 5: From: "Kai-Uwe Jensen";tag=1373ec14 (61) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 6: Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. (66) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 7: CSeq: 1 REGISTER (16) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 8: Expires: 3600 (13) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 10: Supported: eventlist (20) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 11: User-Agent: eyeBeam release 3010n stamp 19039 (45) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 12: Content-Length: 0 (17) Dec 25 13:11:48 DEBUG[786] chan_sip.c: Header 13: (0) Dec 25 13:11:48 VERBOSE[786] logger.c: --- (13 headers 0 lines)Dec 25 13:11:48 VERBOSE[786] logger.c: --- (13 headers 0 lines)--- Dec 25 13:11:48 DEBUG[786] acl.c: ##### Testing 192.168.254.1 with 192.168.254.0 Dec 25 13:11:48 DEBUG[786] chan_sip.c: Allocating new SIP dialog for fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. - REGISTER (No RTP) Dec 25 13:11:48 DEBUG[786] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER Dec 25 13:11:48 VERBOSE[786] logger.c: Using latest REGISTER request as basis request Dec 25 13:11:48 VERBOSE[786] logger.c: Sending to 192.168.254.60 : 8917 (non-NAT) Dec 25 13:11:48 VERBOSE[786] logger.c: Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-233d805fde66af54-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=1373ec14 To: "Kai-Uwe Jensen" Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Dec 25 13:11:48 VERBOSE[786] logger.c: Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-233d805fde66af54-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=1373ec14 To: "Kai-Uwe Jensen";tag=as54db431e Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="4edce5f9" Content-Length: 0 --- Dec 25 13:11:48 VERBOSE[786] logger.c: Scheduling destruction of call 'fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' in 15000 ms Dec 25 13:11:49 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: REGISTER sip:cojensen.net SIP/2.0 Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-1e52b75f1c0aa32d-1--d87543-;rport Max-Forwards: 70 Contact: To: "Kai-Uwe Jensen" From: "Kai-Uwe Jensen";tag=1373ec14 Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Authorization: Digest username="ext2003",realm="asterisk",nonce="4edce5f9",uri="sip:cojensen.net",response="54799cb33262545fac4b493307ad744e",algorithm=MD5 Content-Length: 0 Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 0: REGISTER sip:cojensen.net SIP/2.0 (33) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-1e52b75f1c0aa32d-1--d87543-;rport (92) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 2: Max-Forwards: 70 (16) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 3: Contact: (42) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 4: To: "Kai-Uwe Jensen" (46) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 5: From: "Kai-Uwe Jensen";tag=1373ec14 (61) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 6: Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. (66) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 7: CSeq: 2 REGISTER (16) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 8: Expires: 3600 (13) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 10: Supported: eventlist (20) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 11: User-Agent: eyeBeam release 3010n stamp 19039 (45) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 12: Authorization: Digest username="ext2003",realm="asterisk",nonce="4edce5f9",uri="sip:cojensen.net",response="54799cb33262545fac4b493307ad744e",algorithm=MD5 (155) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 13: Content-Length: 0 (17) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 14: (0) Dec 25 13:11:49 VERBOSE[786] logger.c: --- (14 headers 0 lines)Dec 25 13:11:49 VERBOSE[786] logger.c: --- (14 headers 0 lines)--- Dec 25 13:11:49 DEBUG[786] chan_sip.c: = Found Their Call ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. Their Tag 1373ec14 Our tag: as54db431e Dec 25 13:11:49 DEBUG[786] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER Dec 25 13:11:49 VERBOSE[786] logger.c: Using latest REGISTER request as basis request Dec 25 13:11:49 VERBOSE[786] logger.c: Sending to 192.168.254.60 : 8917 (NAT) Dec 25 13:11:49 VERBOSE[786] logger.c: Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-1e52b75f1c0aa32d-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=1373ec14 To: "Kai-Uwe Jensen" Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Dec 25 13:11:49 DEBUG[786] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Dec 25 13:11:49 DEBUG[786] acl.c: ##### Testing 192.168.254.1 with 192.168.254.0 Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 0: OPTIONS sip:ext2003@192.168.254.60:8917 SIP/2.0 (47) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK16103de9;rport (66) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as629b9798 (62) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 3: To: (37) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 4: Contact: (39) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 5: Call-ID: 725dfd3d58948422755b718a06163e70@192.168.254.250 (57) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 8: Max-Forwards: 70 (16) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 9: Date: Sun, 25 Dec 2005 20:11:49 GMT (35) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 11: Content-Length: 0 (17) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 12: (0) Dec 25 13:11:49 VERBOSE[786] logger.c: 12 headers, 0 lines Dec 25 13:11:49 VERBOSE[786] logger.c: Reliably Transmitting (no NAT) to 192.168.254.1:8917: OPTIONS sip:ext2003@192.168.254.60:8917 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK16103de9;rport From: "Asterisk" ;tag=as629b9798 To: Contact: Call-ID: 725dfd3d58948422755b718a06163e70@192.168.254.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 25 Dec 2005 20:11:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Dec 25 13:11:49 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #45 Dec 25 13:11:49 VERBOSE[786] logger.c: -- Registered SIP 'ext2003' at 192.168.254.1 port 8917 expires 3600 Dec 25 13:11:49 VERBOSE[786] logger.c: -- Saved useragent "eyeBeam release 3010n stamp 19039" for peer ext2003 Dec 25 13:11:49 VERBOSE[786] logger.c: Transmitting (NAT) to 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.60:8917;branch=z9hG4bK-d87543-1e52b75f1c0aa32d-1--d87543-;received=192.168.254.1;rport=8917 From: "Kai-Uwe Jensen";tag=1373ec14 To: "Kai-Uwe Jensen";tag=as54db431e Call-ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 3600 Contact: ;expires=3600 Date: Sun, 25 Dec 2005 20:11:49 GMT Content-Length: 0 --- Dec 25 13:11:49 VERBOSE[786] logger.c: Scheduling destruction of call 'fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' in 15000 ms Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[783] devicestate.c: Changing state for SIP/ext2003 - state 1 (Not in use) Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[795] app_queue.c: Device 'SIP/ext2003' changed to state '1' (Not in use) Dec 25 13:11:49 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK16103de9;rport=5060;received=67.174.106.30 Contact: To: ;tag=573d6b6e From: "Asterisk";tag=as629b9798 Call-ID: 725dfd3d58948422755b718a06163e70@192.168.254.250 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK16103de9;rport=5060;received=67.174.106.30 (94) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 2: Contact: (34) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 3: To: ;tag=573d6b6e (50) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 4: From: "Asterisk";tag=as629b9798 (61) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 5: Call-ID: 725dfd3d58948422755b718a06163e70@192.168.254.250 (57) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 7: Accept: application/sdp (23) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 8: Accept-Language: en (19) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 10: Supported: eventlist (20) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 11: User-Agent: eyeBeam release 3010n stamp 19039 (45) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 12: Content-Length: 0 (17) Dec 25 13:11:49 DEBUG[786] chan_sip.c: Header 13: (0) Dec 25 13:11:49 VERBOSE[786] logger.c: --- (13 headers 0 lines)Dec 25 13:11:49 VERBOSE[786] logger.c: --- (13 headers 0 lines)--- Dec 25 13:11:49 DEBUG[786] chan_sip.c: = Found Their Call ID: 725dfd3d58948422755b718a06163e70@192.168.254.250 Their Tag Our tag: as629b9798 Dec 25 13:11:49 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #45 Dec 25 13:11:49 DEBUG[786] chan_sip.c: Stopping retransmission on '725dfd3d58948422755b718a06163e70@192.168.254.250' of Request 102: Match Found Dec 25 13:11:49 VERBOSE[786] logger.c: Destroying call '725dfd3d58948422755b718a06163e70@192.168.254.250' Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[783] devicestate.c: Changing state for SIP/ext2003 - state 1 (Not in use) Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[783] chan_sip.c: Checking device state for peer ext2003 Dec 25 13:11:49 DEBUG[796] app_queue.c: Device 'SIP/ext2003' changed to state '1' (Not in use) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) Dec 25 13:11:52 DEBUG[786] acl.c: ##### Testing 192.168.254.1 with 192.168.254.0 Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 0: NOTIFY sip:ext2003@192.168.254.60:8917 SIP/2.0 (46) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK67b09443;rport (66) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as4a74068b (62) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 3: To: (37) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 4: Contact: (39) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 5: Call-ID: 2130c388545a22f66e8694282770dda9@192.168.254.250 (57) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 NOTIFY (16) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 8: Max-Forwards: 70 (16) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 9: Event: message-summary (22) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 10: Content-Type: application/simple-message-summary (48) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 11: Content-Length: 80 (18) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 12: (0) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Line: Messages-Waiting: no (20) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Line: Message-Account: sip:asterisk@ (30) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Line: Voice-Message: 0/0 (0/0) (24) Dec 25 13:11:52 VERBOSE[786] logger.c: 12 headers, 3 lines Dec 25 13:11:52 VERBOSE[786] logger.c: Reliably Transmitting (NAT) to 192.168.254.1:8917: NOTIFY sip:ext2003@192.168.254.60:8917 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK67b09443;rport From: "Asterisk" ;tag=as4a74068b To: Contact: Call-ID: 2130c388545a22f66e8694282770dda9@192.168.254.250 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) --- Dec 25 13:11:52 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #49 Dec 25 13:11:52 VERBOSE[786] logger.c: Scheduling destruction of call '2130c388545a22f66e8694282770dda9@192.168.254.250' in 15000 ms Dec 25 13:11:52 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK67b09443;rport=5060;received=67.174.106.30 Contact: To: ;tag=de237a08 From: "Asterisk";tag=as4a74068b Call-ID: 2130c388545a22f66e8694282770dda9@192.168.254.250 CSeq: 102 NOTIFY User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK67b09443;rport=5060;received=67.174.106.30 (94) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 2: Contact: (34) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 3: To: ;tag=de237a08 (50) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 4: From: "Asterisk";tag=as4a74068b (61) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 5: Call-ID: 2130c388545a22f66e8694282770dda9@192.168.254.250 (57) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 NOTIFY (16) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 7: User-Agent: eyeBeam release 3010n stamp 19039 (45) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 8: Content-Length: 0 (17) Dec 25 13:11:52 DEBUG[786] chan_sip.c: Header 9: (0) Dec 25 13:11:52 VERBOSE[786] logger.c: --- (9 headers 0 lines)Dec 25 13:11:52 VERBOSE[786] logger.c: --- (9 headers 0 lines)--- Dec 25 13:11:52 DEBUG[786] chan_sip.c: = Found Their Call ID: 2130c388545a22f66e8694282770dda9@192.168.254.250 Their Tag Our tag: as4a74068b Dec 25 13:11:52 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #49 Dec 25 13:11:52 DEBUG[786] chan_sip.c: Stopping retransmission on '2130c388545a22f66e8694282770dda9@192.168.254.250' of Request 102: Match Found Dec 25 13:11:52 VERBOSE[786] logger.c: Destroying call '2130c388545a22f66e8694282770dda9@192.168.254.250' Dec 25 13:11:57 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.252:5060: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKe66f5b5e2FFFD311 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: CSeq: 1 INVITE Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 243 v=0 o=- 1135541282 1135541282 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 0: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 (50) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKe66f5b5e2FFFD311 (68) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 2: From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 (74) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 3: To: (41) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 5: Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 (51) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 6: Contact: (43) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 (54) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 9: Supported: 100rel,replace (25) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 11: Max-Forwards: 70 (16) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 12: Content-Type: application/sdp (29) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 13: Content-Length: 243 (19) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 14: (0) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: v=0 (3) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: o=- 1135541282 1135541282 IN IP4 192.168.254.252 (48) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: s=Polycom IP Phone (18) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: c=IN IP4 192.168.254.252 (24) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: t=0 0 (5) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: m=audio 2232 RTP/AVP 0 8 18 101 (31) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Dec 25 13:11:57 VERBOSE[786] logger.c: --- (14 headers 10 lines)Dec 25 13:11:57 VERBOSE[786] logger.c: --- (14 headers 10 lines)--- Dec 25 13:11:57 DEBUG[786] chan_sip.c: = No match Their Call ID: fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA.. Their Tag 1373ec14 Our tag: as54db431e Dec 25 13:11:57 DEBUG[786] acl.c: ##### Testing 192.168.254.252 with 192.168.254.0 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Allocating new SIP dialog for 1afa3b78-d953d662-5cf653c3@192.168.254.252 - INVITE (With RTP) Dec 25 13:11:57 DEBUG[786] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Dec 25 13:11:57 DEBUG[786] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replace" Dec 25 13:11:57 DEBUG[786] chan_sip.c: Found SIP option: -100rel- Dec 25 13:11:57 DEBUG[786] chan_sip.c: Matched SIP option: 100rel Dec 25 13:11:57 DEBUG[786] chan_sip.c: Found SIP option: -replace- Dec 25 13:11:57 DEBUG[786] chan_sip.c: Found no match for SIP option: replace (Please file bug report!) Dec 25 13:11:57 DEBUG[786] chan_sip.c: * SIP extension value: 2 for call 1afa3b78-d953d662-5cf653c3@192.168.254.252 Dec 25 13:11:57 VERBOSE[786] logger.c: Using INVITE request as basis request - 1afa3b78-d953d662-5cf653c3@192.168.254.252 Dec 25 13:11:57 VERBOSE[786] logger.c: Sending to 192.168.254.252 : 5060 (non-NAT) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Setting NAT on RTP to 0 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Setting NAT on VRTP to 0 Dec 25 13:11:57 VERBOSE[786] logger.c: Reliably Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKe66f5b5e2FFFD311;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: ;tag=as6eeeb5a8 Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="1aa2a79f" Content-Length: 0 --- Dec 25 13:11:57 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #51 Dec 25 13:11:57 VERBOSE[786] logger.c: Scheduling destruction of call '1afa3b78-d953d662-5cf653c3@192.168.254.252' in 15000 ms Dec 25 13:11:57 VERBOSE[786] logger.c: Found user 'ext2006' Dec 25 13:11:57 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 0: (0) Dec 25 13:11:57 VERBOSE[786] logger.c: --- (0 headers 0 lines)Dec 25 13:11:57 VERBOSE[786] logger.c: --- (0 headers 0 lines) Nat keepalive Dec 25 13:11:57 VERBOSE[786] logger.c: --- (0 headers 0 lines) Nat keepalive --- Dec 25 13:11:57 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.252:5060: ACK sip:2600@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKe66f5b5e2FFFD311 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: ;tag=as6eeeb5a8 CSeq: 1 ACK Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 0: ACK sip:2600@192.168.254.250 SIP/2.0 (36) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKe66f5b5e2FFFD311 (68) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 2: From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 (74) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 3: To: ;tag=as6eeeb5a8 (56) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 4: CSeq: 1 ACK (11) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 5: Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 (51) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 6: Contact: (43) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 (54) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 9: Max-Forwards: 70 (16) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 10: Content-Length: 0 (17) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 11: (0) Dec 25 13:11:57 VERBOSE[786] logger.c: --- (11 headers 0 lines)Dec 25 13:11:57 VERBOSE[786] logger.c: --- (11 headers 0 lines)--- Dec 25 13:11:57 DEBUG[786] chan_sip.c: = Found Their Call ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Their Tag 8509D29C-E16AAC05 Our tag: as6eeeb5a8 Dec 25 13:11:57 DEBUG[786] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Dec 25 13:11:57 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #51 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Stopping retransmission on '1afa3b78-d953d662-5cf653c3@192.168.254.252' of Response 1: Match Found Dec 25 13:11:57 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.252:5060: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK7c8c28578BA58AA6 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: CSeq: 2 INVITE Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="ext2006", realm="asterisk", nonce="1aa2a79f", uri="sip:2600@192.168.254.250;user=phone", response="124b06ebf8dba9bc4611e1f001d1e9ab", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 243 v=0 o=- 1135541282 1135541282 IN IP4 192.168.254.252 s=Polycom IP Phone c=IN IP4 192.168.254.252 t=0 0 m=audio 2232 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 0: INVITE sip:2600@192.168.254.250;user=phone SIP/2.0 (50) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK7c8c28578BA58AA6 (68) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 2: From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 (74) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 3: To: (41) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 4: CSeq: 2 INVITE (14) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 5: Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 (51) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 6: Contact: (43) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 (54) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 9: Supported: 100rel,replace (25) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 11: Proxy-Authorization: Digest username="ext2006", realm="asterisk", nonce="1aa2a79f", uri="sip:2600@192.168.254.250;user=phone", response="124b06ebf8dba9bc4611e1f001d1e9ab", algorithm=MD5 (185) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 12: Max-Forwards: 70 (16) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 13: Content-Type: application/sdp (29) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 14: Content-Length: 243 (19) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Header 15: (0) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: v=0 (3) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: o=- 1135541282 1135541282 IN IP4 192.168.254.252 (48) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: s=Polycom IP Phone (18) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: c=IN IP4 192.168.254.252 (24) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: t=0 0 (5) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: m=audio 2232 RTP/AVP 0 8 18 101 (31) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Dec 25 13:11:57 VERBOSE[786] logger.c: --- (15 headers 10 lines)Dec 25 13:11:57 VERBOSE[786] logger.c: --- (15 headers 10 lines)--- Dec 25 13:11:57 DEBUG[786] chan_sip.c: = Found Their Call ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Their Tag 8509D29C-E16AAC05 Our tag: as6eeeb5a8 Dec 25 13:11:57 DEBUG[786] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Dec 25 13:11:57 VERBOSE[786] logger.c: Using INVITE request as basis request - 1afa3b78-d953d662-5cf653c3@192.168.254.252 Dec 25 13:11:57 VERBOSE[786] logger.c: Sending to 192.168.254.252 : 5060 (non-NAT) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Setting NAT on RTP to 0 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Setting NAT on VRTP to 0 Dec 25 13:11:57 VERBOSE[786] logger.c: Found user 'ext2006' Dec 25 13:11:57 VERBOSE[786] logger.c: Found RTP audio format 0 Dec 25 13:11:57 VERBOSE[786] logger.c: Found RTP audio format 8 Dec 25 13:11:57 VERBOSE[786] logger.c: Found RTP audio format 18 Dec 25 13:11:57 VERBOSE[786] logger.c: Found RTP audio format 101 Dec 25 13:11:57 VERBOSE[786] logger.c: Peer audio RTP is at port 192.168.254.252:2232 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Peer audio RTP is at port 192.168.254.252:2232 Dec 25 13:11:57 VERBOSE[786] logger.c: Peer video RTP is at port 192.168.254.252:65535 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Peer video RTP is at port 192.168.254.252:65535 Dec 25 13:11:57 VERBOSE[786] logger.c: Found description format PCMU Dec 25 13:11:57 VERBOSE[786] logger.c: Found description format PCMA Dec 25 13:11:57 VERBOSE[786] logger.c: Found description format G729 Dec 25 13:11:57 VERBOSE[786] logger.c: Found description format telephone-event Dec 25 13:11:57 VERBOSE[786] logger.c: Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Dec 25 13:11:57 VERBOSE[786] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Dec 25 13:11:57 DEBUG[786] chan_sip.c: Checking SIP call limits for device ext2006 Dec 25 13:11:57 DEBUG[786] chan_sip.c: Updating call counter for incoming call Dec 25 13:11:57 VERBOSE[786] logger.c: Looking for 2600 in home (domain 192.168.254.250) Dec 25 13:11:57 DEBUG[786] chan_sip.c: build_route: Contact hop: Dec 25 13:11:57 VERBOSE[786] logger.c: list_route: hop: Dec 25 13:11:57 VERBOSE[786] logger.c: Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK7c8c28578BA58AA6;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[783] devicestate.c: Changing state for SIP/ext2006 - state 2 (In use) Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[797] pbx.c: Launching 'Goto' Dec 25 13:11:57 VERBOSE[797] logger.c: -- Executing Goto("SIP/ext2006-5563", "confline|s|1") in new stack Dec 25 13:11:57 VERBOSE[797] logger.c: -- Goto (confline,s,1) Dec 25 13:11:57 DEBUG[797] pbx.c: Launching 'Answer' Dec 25 13:11:57 VERBOSE[797] logger.c: -- Executing Answer("SIP/ext2006-5563", "") in new stack Dec 25 13:11:57 DEBUG[797] chan_sip.c: sip_answer(SIP/ext2006-5563) Dec 25 13:11:57 VERBOSE[797] logger.c: We're at 192.168.254.250 port 9038 Dec 25 13:11:57 VERBOSE[797] logger.c: Video is at 192.168.254.250 port 9040 Dec 25 13:11:57 VERBOSE[797] logger.c: Adding codec 0x4 (ulaw) to SDP Dec 25 13:11:57 VERBOSE[797] logger.c: Adding codec 0x100 (g729) to SDP Dec 25 13:11:57 VERBOSE[797] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Dec 25 13:11:57 VERBOSE[797] logger.c: Reliably Transmitting (no NAT) to 192.168.254.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bK7c8c28578BA58AA6;received=192.168.254.252 From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: ;tag=as62828bee Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=root 732 732 IN IP4 192.168.254.250 s=session c=IN IP4 192.168.254.250 t=0 0 m=audio 9038 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Dec 25 13:11:57 DEBUG[797] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #53 Dec 25 13:11:57 DEBUG[797] pbx.c: Launching 'Wait' Dec 25 13:11:57 VERBOSE[797] logger.c: -- Executing Wait("SIP/ext2006-5563", "1") in new stack Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[783] devicestate.c: Changing state for SIP/ext2006 - state 2 (In use) Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[783] chan_sip.c: Checking device state for peer ext2006 Dec 25 13:11:57 DEBUG[798] app_queue.c: Device 'SIP/ext2006' changed to state '2' (In use) Dec 25 13:11:57 DEBUG[799] app_queue.c: Device 'SIP/ext2006' changed to state '2' (In use) Dec 25 13:11:58 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.252:5060: ACK sip:2600@192.168.254.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKa4b4dc2a55538EAD From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 To: ;tag=as62828bee CSeq: 2 ACK Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Max-Forwards: 70 Content-Length: 0 Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 0: ACK sip:2600@192.168.254.250 SIP/2.0 (36) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.252:5060;branch=z9hG4bKa4b4dc2a55538EAD (68) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 2: From: "Kai-Uwe Jensen" ;tag=8509D29C-E16AAC05 (74) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 3: To: ;tag=as62828bee (56) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 4: CSeq: 2 ACK (11) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 5: Call-ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 (51) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 6: Contact: (43) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 (54) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 9: Max-Forwards: 70 (16) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 10: Content-Length: 0 (17) Dec 25 13:11:58 DEBUG[786] chan_sip.c: Header 11: (0) Dec 25 13:11:58 VERBOSE[786] logger.c: --- (11 headers 0 lines)Dec 25 13:11:58 VERBOSE[786] logger.c: --- (11 headers 0 lines)--- Dec 25 13:11:58 DEBUG[786] chan_sip.c: = Found Their Call ID: 1afa3b78-d953d662-5cf653c3@192.168.254.252 Their Tag 8509D29C-E16AAC05 Our tag: as62828bee Dec 25 13:11:58 DEBUG[786] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Dec 25 13:11:58 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #53 Dec 25 13:11:58 DEBUG[786] chan_sip.c: Stopping retransmission on '1afa3b78-d953d662-5cf653c3@192.168.254.252' of Response 2: Match Found Dec 25 13:11:58 DEBUG[788] chan_iax2.c: Allocate call number Dec 25 13:11:58 DEBUG[788] chan_iax2.c: Registration created on call 2 Dec 25 13:11:58 DEBUG[797] pbx.c: Launching 'BackGround' Dec 25 13:11:58 VERBOSE[797] logger.c: -- Executing BackGround("SIP/ext2006-5563", "enter-conf-call-number") in new stack Dec 25 13:11:58 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format gsm Dec 25 13:11:59 DEBUG[797] rtp.c: Ooh, format changed from unknown to ulaw Dec 25 13:11:59 DEBUG[797] channel.c: Scheduling timer at 160 sample intervals Dec 25 13:11:59 VERBOSE[797] logger.c: -- Playing 'enter-conf-call-number' (language 'en') Dec 25 13:12:00 DEBUG[797] rtp.c: Sending dtmf: 56 (8), at 192.168.254.252 Dec 25 13:12:00 DEBUG[797] channel.c: Scheduling timer at 0 sample intervals Dec 25 13:12:00 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format ulaw Dec 25 13:12:00 DEBUG[797] pbx.c: Oooh, got something to jump out with ('8')! Dec 25 13:12:00 DEBUG[797] rtp.c: Sending dtmf: 57 (9), at 192.168.254.252 Dec 25 13:12:00 DEBUG[797] rtp.c: Sending dtmf: 56 (8), at 192.168.254.252 Dec 25 13:12:00 DEBUG[797] rtp.c: Sending dtmf: 55 (7), at 192.168.254.252 Dec 25 13:12:01 DEBUG[797] rtp.c: Sending dtmf: 55 (7), at 192.168.254.252 Dec 25 13:12:01 DEBUG[797] rtp.c: Sending dtmf: 54 (6), at 192.168.254.252 Dec 25 13:12:01 DEBUG[797] rtp.c: Sending dtmf: 50 (2), at 192.168.254.252 Dec 25 13:12:02 DEBUG[797] rtp.c: Sending dtmf: 35 (#), at 192.168.254.252 Dec 25 13:12:02 VERBOSE[797] logger.c: == CDR updated on SIP/ext2006-5563 Dec 25 13:12:02 DEBUG[797] pbx.c: Launching 'Goto' Dec 25 13:12:02 VERBOSE[797] logger.c: -- Executing Goto("SIP/ext2006-5563", "8987762|1") in new stack Dec 25 13:12:02 VERBOSE[797] logger.c: -- Goto (confline,8987762,1) Dec 25 13:12:02 DEBUG[797] pbx.c: Launching 'Wait' Dec 25 13:12:02 VERBOSE[797] logger.c: -- Executing Wait("SIP/ext2006-5563", "1") in new stack Dec 25 13:12:03 DEBUG[797] pbx.c: Launching 'MeetMe' Dec 25 13:12:03 VERBOSE[797] logger.c: -- Executing MeetMe("SIP/ext2006-5563", "8987762|MPxwsvn") in new stack Dec 25 13:12:03 DEBUG[797] config.c: Parsing /etc/asterisk/meetme.conf Dec 25 13:12:03 DEBUG[797] chan_zap.c: Using channel -2 Dec 25 13:12:03 DEBUG[797] channel.c: Set channel Zap/pseudo-2101231415 to read format slin Dec 25 13:12:03 DEBUG[797] channel.c: Set channel Zap/pseudo-2101231415 to write format slin Dec 25 13:12:03 VERBOSE[797] logger.c: -- Created MeetMe conference 1023 for conference '8987762' Dec 25 13:12:03 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format gsm Dec 25 13:12:03 DEBUG[797] rtp.c: Difference is 24192, ms is 3044 Dec 25 13:12:03 DEBUG[797] channel.c: Scheduling timer at 160 sample intervals Dec 25 13:12:03 VERBOSE[797] logger.c: -- Playing 'conf-waitforleader' (language 'en') Dec 25 13:12:03 DEBUG[783] devicestate.c: Changing state for Zap/pseudo - state 2 (In use) Dec 25 13:12:03 DEBUG[801] app_queue.c: Device 'Zap/pseudo' changed to state '2' (In use) Dec 25 13:12:04 DEBUG[786] chan_sip.c: Auto destroying call 'fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' Dec 25 13:12:04 VERBOSE[786] logger.c: Destroying call 'fb3328517b393d20@TUFETUFOMi5hbWVyaWNhcy5ocHFjb3JwLm5ldA..' Dec 25 13:12:05 DEBUG[797] channel.c: Scheduling timer at 0 sample intervals Dec 25 13:12:05 DEBUG[797] channel.c: Scheduling timer at 0 sample intervals Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format ulaw Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format slin Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to read format slin Dec 25 13:12:05 DEBUG[797] app_meetme.c: Placed channel SIP/ext2006-5563 in ZAP conf 1023 Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format slin Dec 25 13:12:05 VERBOSE[797] logger.c: -- Started music on hold, class 'default', on SIP/ext2006-5563 Dec 25 13:12:05 DEBUG[797] channel.c: Scheduling timer at 160 sample intervals Dec 25 13:12:05 DEBUG[797] channel.c: Generator got voice, switching to phase locked mode Dec 25 13:12:05 DEBUG[797] channel.c: Scheduling timer at 0 sample intervals Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format slin Dec 25 13:12:05 DEBUG[797] channel.c: Set channel SIP/ext2006-5563 to write format slin Dec 25 13:12:05 DEBUG[797] res_musiconhold.c: SIP/ext2006-5563 Opened file 4 '/var/lib/asterisk/mohmp3/MoH' Dec 25 13:12:06 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.1:8917: Dec 25 13:12:06 DEBUG[786] chan_sip.c: Header 0: (0) Dec 25 13:12:06 VERBOSE[786] logger.c: --- (0 headers 0 lines)Dec 25 13:12:06 VERBOSE[786] logger.c: --- (0 headers 0 lines) Nat keepalive Dec 25 13:12:06 VERBOSE[786] logger.c: --- (0 headers 0 lines) Nat keepalive --- Dec 25 13:12:07 DEBUG[797] rtp.c: Got RTCP report of 92 bytes Dec 25 13:12:07 DEBUG[797] app_meetme.c: Got unrecognized frame on channel SIP/ext2006-5563, f->frametype=5,f->subclass=0 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Dec 25 13:12:08 DEBUG[786] acl.c: ##### Testing 192.168.254.251 with 192.168.254.0 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: OPTIONS sip:ext2001@192.168.254.251:5061 SIP/2.0 (48) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK7358539e;rport (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as74dd2da1 (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: To: (38) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: Contact: (39) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Call-ID: 5ffb5af4554739c24952fe2b10fcb058@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: Max-Forwards: 70 (16) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Date: Sun, 25 Dec 2005 20:12:08 GMT (35) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 11: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 12: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: 12 headers, 0 lines Dec 25 13:12:08 VERBOSE[786] logger.c: Reliably Transmitting (no NAT) to 192.168.254.251:5061: OPTIONS sip:ext2001@192.168.254.251:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK7358539e;rport From: "Asterisk" ;tag=as74dd2da1 To: Contact: Call-ID: 5ffb5af4554739c24952fe2b10fcb058@192.168.254.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 25 Dec 2005 20:12:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Dec 25 13:12:08 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #54 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Dec 25 13:12:08 DEBUG[786] acl.c: ##### Testing 192.168.254.251 with 192.168.254.0 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: OPTIONS sip:ext2002@192.168.254.251:5062 SIP/2.0 (48) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK793a5aba;rport (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as28800667 (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: To: (38) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: Contact: (39) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Call-ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: Max-Forwards: 70 (16) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Date: Sun, 25 Dec 2005 20:12:08 GMT (35) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 11: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 12: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: 12 headers, 0 lines Dec 25 13:12:08 VERBOSE[786] logger.c: Reliably Transmitting (no NAT) to 192.168.254.251:5062: OPTIONS sip:ext2002@192.168.254.251:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK793a5aba;rport From: "Asterisk" ;tag=as28800667 To: Contact: Call-ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 25 Dec 2005 20:12:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Dec 25 13:12:08 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #56 Dec 25 13:12:08 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.251:5061: SIP/2.0 200 OK To: ;tag=f81c8eb560b46b85i0 From: "Asterisk" ;tag=as74dd2da1 Call-ID: 5ffb5af4554739c24952fe2b10fcb058@192.168.254.250 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK7358539e;rport=5060 Server: Linksys/PAP2-3.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: To: ;tag=f81c8eb560b46b85i0 (61) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as74dd2da1 (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: Call-ID: 5ffb5af4554739c24952fe2b10fcb058@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK7358539e;rport=5060 (71) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: Server: Linksys/PAP2-3.1.6(LS) (30) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Supported: x-sipura (19) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)--- Dec 25 13:12:08 DEBUG[786] chan_sip.c: = No match Their Call ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 Their Tag Our tag: as28800667 Dec 25 13:12:08 DEBUG[786] chan_sip.c: = Found Their Call ID: 5ffb5af4554739c24952fe2b10fcb058@192.168.254.250 Their Tag Our tag: as74dd2da1 Dec 25 13:12:08 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #54 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Stopping retransmission on '5ffb5af4554739c24952fe2b10fcb058@192.168.254.250' of Request 102: Match Found Dec 25 13:12:08 VERBOSE[786] logger.c: Destroying call '5ffb5af4554739c24952fe2b10fcb058@192.168.254.250' Dec 25 13:12:08 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.251:5062: SIP/2.0 200 OK To: ;tag=f9bf535cb983705i1 From: "Asterisk" ;tag=as28800667 Call-ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK793a5aba;rport=5060 Server: Linksys/PAP2-3.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: To: ;tag=f9bf535cb983705i1 (60) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as28800667 (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: Call-ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK793a5aba;rport=5060 (71) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: Server: Linksys/PAP2-3.1.6(LS) (30) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Supported: x-sipura (19) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)--- Dec 25 13:12:08 DEBUG[786] chan_sip.c: = Found Their Call ID: 75357bae4258eb710996f3685f2f1a0d@192.168.254.250 Their Tag Our tag: as28800667 Dec 25 13:12:08 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #56 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Stopping retransmission on '75357bae4258eb710996f3685f2f1a0d@192.168.254.250' of Request 102: Match Found Dec 25 13:12:08 VERBOSE[786] logger.c: Destroying call '75357bae4258eb710996f3685f2f1a0d@192.168.254.250' Dec 25 13:12:08 DEBUG[786] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Dec 25 13:12:08 DEBUG[786] acl.c: ##### Testing 192.168.254.252 with 192.168.254.0 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: OPTIONS sip:ext2006@192.168.254.252:5060 SIP/2.0 (48) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK36fde707;rport (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as518b46bb (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: To: (38) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: Contact: (39) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Call-ID: 19652cc4201604d1093aa8ef3b425828@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: Max-Forwards: 70 (16) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Date: Sun, 25 Dec 2005 20:12:08 GMT (35) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 11: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 12: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: 12 headers, 0 lines Dec 25 13:12:08 VERBOSE[786] logger.c: Reliably Transmitting (no NAT) to 192.168.254.252:5060: OPTIONS sip:ext2006@192.168.254.252:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK36fde707;rport From: "Asterisk" ;tag=as518b46bb To: Contact: Call-ID: 19652cc4201604d1093aa8ef3b425828@192.168.254.250 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 25 Dec 2005 20:12:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Dec 25 13:12:08 DEBUG[786] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #60 Dec 25 13:12:08 VERBOSE[786] logger.c: <-- SIP read from 192.168.254.252:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK36fde707;rport From: "Asterisk" ;tag=as518b46bb To: ;tag=C8796EEE-34E1483F CSeq: 102 OPTIONS Call-ID: 19652cc4201604d1093aa8ef3b425828@192.168.254.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 Content-Length: 0 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.254.250:5060;branch=z9hG4bK36fde707;rport (66) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 2: From: "Asterisk" ;tag=as518b46bb (62) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 3: To: ;tag=C8796EEE-34E1483F (60) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 5: Call-ID: 19652cc4201604d1093aa8ef3b425828@192.168.254.250 (57) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 6: Contact: (43) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.6.2.0041 (54) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 9: Content-Length: 0 (17) Dec 25 13:12:08 DEBUG[786] chan_sip.c: Header 10: (0) Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)Dec 25 13:12:08 VERBOSE[786] logger.c: --- (10 headers 0 lines)--- Dec 25 13:12:08 DEBUG[786] chan_sip.c: = Found Their Call ID: 19652cc4201604d1093aa8ef3b425828@192.168.254.250 Their Tag Our tag: as518b46bb Dec 25 13:12:08 DEBUG[786] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #60 Dec 25 13:12:08 DEBUG[786] chan_sip.c: Stopping retransmission on '19652cc4201604d1093aa8ef3b425828@192.168.254.250' of Request 102: Match Found Dec 25 13:12:08 VERBOSE[786] logger.c: Destroying call '19652cc4201604d1093aa8ef3b425828@192.168.254.250' Dec 25 13:12:11 DEBUG[800] chan_zap.c: DTMF digit: 2 on Zap/pseudo-2101231415 ################################################################################ Crash here, with automatic restart by "safe_asterisk" ################################################################################ Dec 25 13:12:16 VERBOSE[846] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Dec 25 13:12:16 VERBOSE[846] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': Dec 25 13:12:16 VERBOSE[846] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': Found Dec 25 13:12:16 NOTICE[846] dnsmgr.c: Managed DNS entries will be refreshed every 300 seconds. Dec 25 13:12:16 VERBOSE[846] logger.c: Asterisk Dynamic Loader loading preload modules: ...