[Nov 4 19:50:15] VERBOSE[1815] logger.c: Asterisk Event Logger restarted [Nov 4 19:50:15] VERBOSE[1815] logger.c: Asterisk Queue Logger restarted [Nov 4 19:50:20] DEBUG[1644] acl.c: ##### Testing 192.168.10.10 with 192.168.10.0 [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. - REGISTER (No RTP) [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: = Found Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Nov 4 19:50:20] DEBUG[1644] acl.c: ##### Testing 192.168.10.10 with 192.168.10.0 [Nov 4 19:50:20] DEBUG[1644] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:20] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:20] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:20] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 19:50:20] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: = Found Their Call ID: 57aa198c364a96d14d5d20b82e695001@192.168.10.181 Their Tag Our tag: as732a1ccd [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: Stopping retransmission on '57aa198c364a96d14d5d20b82e695001@192.168.10.181' of Request 102: Match Found [Nov 4 19:50:20] VERBOSE[1644] logger.c: Really destroying SIP dialog '57aa198c364a96d14d5d20b82e695001@192.168.10.181' Method: OPTIONS [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: = No match Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:20] DEBUG[1644] acl.c: ##### Testing 192.168.10.154 with 192.168.10.0 [Nov 4 19:50:20] DEBUG[1644] chan_sip.c: Invalid SIP message - rejected , no callid, len 349 [Nov 4 19:50:26] DEBUG[1644] chan_sip.c: = No match Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:26] DEBUG[1644] acl.c: ##### Testing 192.168.10.163 with 192.168.10.0 [Nov 4 19:50:26] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for d585d491-dc048da7-e478da@192.168.10.163 - REGISTER (No RTP) [Nov 4 19:50:26] DEBUG[1644] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:50:26] DEBUG[1644] chan_sip.c: = Found Their Call ID: d585d491-dc048da7-e478da@192.168.10.163 Their Tag 955EBB4D-7EA49688 Our tag: as0e622a65 [Nov 4 19:50:26] DEBUG[1644] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 4 19:50:26] DEBUG[1644] devicestate.c: Notification of state change to be queued on device/channel SIP/623 [Nov 4 19:50:26] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 19:50:26] DEBUG[1641] chan_sip.c: Checking device state for peer 623 [Nov 4 19:50:26] DEBUG[1641] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 19:50:26] DEBUG[1662] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 19:50:27] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Nov 4 19:50:27] DEBUG[1644] acl.c: ##### Testing 192.168.10.163 with 192.168.10.0 [Nov 4 19:50:27] DEBUG[1644] chan_sip.c: = Found Their Call ID: 190043a45af2bc655eae6d5b05103892@192.168.10.181 Their Tag Our tag: as32ef4dc5 [Nov 4 19:50:27] DEBUG[1644] chan_sip.c: Stopping retransmission on '190043a45af2bc655eae6d5b05103892@192.168.10.181' of Request 102: Match Found [Nov 4 19:50:27] VERBOSE[1644] logger.c: Really destroying SIP dialog '190043a45af2bc655eae6d5b05103892@192.168.10.181' Method: NOTIFY [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Nov 4 19:50:29] DEBUG[1644] acl.c: ##### Testing 192.168.10.10 with 192.168.10.0 [Nov 4 19:50:29] VERBOSE[1644] logger.c: Scheduling destruction of SIP dialog '5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181' in 6976 ms (Method: NOTIFY) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 0: NOTIFY sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 (69) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK2e0c955b;rport (65) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 2: From: "asterisk" ;tag=as573813c8 (61) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 3: To: (60) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 4: Contact: (38) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181 (56) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 6: CSeq: 102 NOTIFY (16) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 9: Event: message-summary (22) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 10: Content-Type: application/simple-message-summary (48) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 11: Content-Length: 95 (18) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 12: (0) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Line: Messages-Waiting: yes (21) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Line: Message-Account: sip:asterisk@192.168.10.181 (44) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Line: Voice-Message: 5/0 (0/0) (24) [Nov 4 19:50:29] VERBOSE[1644] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:56602: NOTIFY sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK2e0c955b;rport From: "asterisk" ;tag=as573813c8 To: Contact: Call-ID: 5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 95 Messages-Waiting: yes Message-Account: sip:asterisk@192.168.10.181 Voice-Message: 5/0 (0/0) --- [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 4 19:50:29] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.10:56602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK2e0c955b;rport=5060 Contact: To: ;tag=9908c675 From: "asterisk";tag=as573813c8 Call-ID: 5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181 CSeq: 102 NOTIFY Content-Length: 0 <-------------> [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK2e0c955b;rport=5060 (70) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 2: Contact: (34) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 3: To: ;tag=9908c675 (73) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 4: From: "asterisk";tag=as573813c8 (60) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181 (56) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 6: CSeq: 102 NOTIFY (16) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 7: Content-Length: 0 (17) [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Header 8: (0) [Nov 4 19:50:29] VERBOSE[1644] logger.c: --- (8 headers 0 lines) --- [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: = Found Their Call ID: 5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181 Their Tag Our tag: as573813c8 [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #863 [Nov 4 19:50:29] DEBUG[1644] chan_sip.c: Stopping retransmission on '5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181' of Request 102: Match Found [Nov 4 19:50:29] VERBOSE[1644] logger.c: Really destroying SIP dialog '5a5474240809ab0c50df9ae6732dbb0d@192.168.10.181' Method: NOTIFY [Nov 4 19:50:32] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> <-------------> [Nov 4 19:50:32] DEBUG[1644] chan_sip.c: Header 0: (0) [Nov 4 19:50:32] DEBUG[1644] chan_sip.c: Line: (0) [Nov 4 19:50:34] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> INVITE sip:500@192.168.10.181;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK750d09cf5D0B3250 From: "623" ;tag=92BC5B1-A620971C To: CSeq: 1 INVITE Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 254 v=0 o=- 1288886352 1288886352 IN IP4 192.168.10.163 s=Polycom IP Phone c=IN IP4 192.168.10.163 t=0 0 m=audio 10018 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 0: INVITE sip:500@192.168.10.181;user=phone SIP/2.0 (48) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK750d09cf5D0B3250 (62) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 2: From: "623" ;tag=92BC5B1-A620971C (57) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 3: To: (39) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 4: CSeq: 1 INVITE (14) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 (50) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 6: Contact: (33) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 (54) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 9: Supported: 100rel,replaces (26) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 11: Max-Forwards: 70 (16) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 13: Content-Length: 254 (19) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 14: (0) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: v=0 (3) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: o=- 1288886352 1288886352 IN IP4 192.168.10.163 (47) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: s=Polycom IP Phone (18) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: c=IN IP4 192.168.10.163 (23) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: t=0 0 (5) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: m=audio 10018 RTP/AVP 0 8 18 101 (32) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=sendrecv (10) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 4 19:50:34] VERBOSE[1644] logger.c: --- (14 headers 11 lines) --- [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: = No match Their Call ID: d585d491-dc048da7-e478da@192.168.10.163 Their Tag 955EBB4D-7EA49688 Our tag: as0e622a65 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: = No match Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:34] DEBUG[1644] acl.c: ##### Testing 192.168.10.163 with 192.168.10.0 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Allocating new SIP dialog for 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 - INVITE (With RTP) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Found SIP option: -100rel- [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Matched SIP option: 100rel [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Found SIP option: -replaces- [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Matched SIP option: replaces [Nov 4 19:50:34] VERBOSE[1644] logger.c: Sending to 192.168.10.163 : 5060 (no NAT) [Nov 4 19:50:34] VERBOSE[1644] logger.c: Using INVITE request as basis request - 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:50:34] VERBOSE[1644] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK750d09cf5D0B3250;received=192.168.10.163 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as4c619ddf Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="146ab11d" Content-Length: 0 <------------> [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 4 19:50:34] VERBOSE[1644] logger.c: Scheduling destruction of SIP dialog '59ba5135-93a9d0ab-e25cb58e@192.168.10.163' in 32000 ms (Method: INVITE) [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found user '623' [Nov 4 19:50:34] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> ACK sip:500@192.168.10.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK750d09cf5D0B3250 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as4c619ddf CSeq: 1 ACK Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 0: ACK sip:500@192.168.10.181 SIP/2.0 (34) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK750d09cf5D0B3250 (62) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 2: From: "623" ;tag=92BC5B1-A620971C (57) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 3: To: ;tag=as4c619ddf (54) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 4: CSeq: 1 ACK (11) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 (50) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 6: Contact: (33) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 (54) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 10: Content-Length: 0 (17) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 11: (0) [Nov 4 19:50:34] VERBOSE[1644] logger.c: --- (11 headers 0 lines) --- [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: = Found Their Call ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Their Tag 92BC5B1-A620971C Our tag: as4c619ddf [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #864 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Stopping retransmission on '59ba5135-93a9d0ab-e25cb58e@192.168.10.163' of Response 1: Match Found [Nov 4 19:50:34] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> INVITE sip:500@192.168.10.181;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 From: "623" ;tag=92BC5B1-A620971C To: CSeq: 2 INVITE Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="18a4d417ef54e647d9421c395f48abcb", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 254 v=0 o=- 1288886352 1288886352 IN IP4 192.168.10.163 s=Polycom IP Phone c=IN IP4 192.168.10.163 t=0 0 m=audio 10018 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 0: INVITE sip:500@192.168.10.181;user=phone SIP/2.0 (48) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 (61) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 2: From: "623" ;tag=92BC5B1-A620971C (57) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 3: To: (39) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 4: CSeq: 2 INVITE (14) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 (50) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 6: Contact: (33) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 (54) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 9: Supported: 100rel,replaces (26) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 11: Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="18a4d417ef54e647d9421c395f48abcb", algorithm=MD5 (179) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 12: Max-Forwards: 70 (16) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 13: Content-Type: application/sdp (29) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 14: Content-Length: 254 (19) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Header 15: (0) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: v=0 (3) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: o=- 1288886352 1288886352 IN IP4 192.168.10.163 (47) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: s=Polycom IP Phone (18) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: c=IN IP4 192.168.10.163 (23) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: t=0 0 (5) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: m=audio 10018 RTP/AVP 0 8 18 101 (32) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=sendrecv (10) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 4 19:50:34] VERBOSE[1644] logger.c: --- (15 headers 11 lines) --- [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: = Found Their Call ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Their Tag 92BC5B1-A620971C Our tag: as4c619ddf [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 4 19:50:34] VERBOSE[1644] logger.c: Sending to 192.168.10.163 : 5060 (no NAT) [Nov 4 19:50:34] VERBOSE[1644] logger.c: Using INVITE request as basis request - 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found user '623' [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing session-level SDP o=- 1288886352 1288886352 IN IP4 192.168.10.163... UNSUPPORTED. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing session-level SDP s=Polycom IP Phone... UNSUPPORTED. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.10.163... OK. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found RTP audio format 0 [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found RTP audio format 8 [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found RTP audio format 18 [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found RTP audio format 101 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found audio description format PCMU for ID 0 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found audio description format PCMA for ID 8 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found audio description format G729 for ID 18 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Nov 4 19:50:34] VERBOSE[1644] logger.c: Found audio description format telephone-event for ID 101 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: T38 state changed to 0 on channel [Nov 4 19:50:34] VERBOSE[1644] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 4 19:50:34] VERBOSE[1644] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0) [Nov 4 19:50:34] VERBOSE[1644] logger.c: Peer audio RTP is at port 192.168.10.163:10018 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: We're settling with these formats: 0xc (ulaw|alaw) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Checking SIP call limits for device 623 [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Updating call counter for incoming call [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: Call from user '623' is 1 out of 2 [Nov 4 19:50:34] VERBOSE[1644] logger.c: Looking for 500 in internal (domain 192.168.10.181) [Nov 4 19:50:34] DEBUG[1644] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Nov 4 19:50:34] DEBUG[1644] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: build_route: Contact hop: [Nov 4 19:50:34] VERBOSE[1644] logger.c: list_route: hop: [Nov 4 19:50:34] DEBUG[1644] chan_sip.c: SIP/623-00000012: New call is still down.... Trying... [Nov 4 19:50:34] VERBOSE[1644] logger.c: <--- Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7;received=192.168.10.163 From: "623" ;tag=92BC5B1-A620971C To: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Length: 0 <------------> [Nov 4 19:50:34] DEBUG[1644] devicestate.c: Notification of state change to be queued on device/channel SIP/623 [Nov 4 19:50:34] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 19:50:34] DEBUG[1641] chan_sip.c: Checking device state for peer 623 [Nov 4 19:50:34] DEBUG[1641] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 19:50:34] DEBUG[1662] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 19:50:34] DEBUG[1816] pbx.c: Launching 'Goto' [Nov 4 19:50:34] VERBOSE[1816] logger.c: -- Executing [500@internal:1] Goto("SIP/623-00000012", "do_support|s|1") in new stack [Nov 4 19:50:34] VERBOSE[1816] logger.c: -- Goto (do_support,s,1) [Nov 4 19:50:34] DEBUG[1816] pbx.c: Launching 'Wait' [Nov 4 19:50:34] VERBOSE[1816] logger.c: -- Executing [s@do_support:1] Wait("SIP/623-00000012", "0.5") in new stack [Nov 4 19:50:35] DEBUG[1816] pbx.c: Launching 'Set' [Nov 4 19:50:35] VERBOSE[1816] logger.c: -- Executing [s@do_support:2] Set("SIP/623-00000012", "CALLERID(name)=MR X") in new stack [Nov 4 19:50:35] DEBUG[1816] pbx.c: Launching 'Set' [Nov 4 19:50:35] VERBOSE[1816] logger.c: -- Executing [s@do_support:3] Set("SIP/623-00000012", "CALLERID(num)=12345") in new stack [Nov 4 19:50:35] DEBUG[1816] pbx.c: Launching 'Queue' [Nov 4 19:50:35] VERBOSE[1816] logger.c: -- Executing [s@do_support:4] Queue("SIP/623-00000012", "queue_support|tT|||1800") in new stack [Nov 4 19:50:35] DEBUG[1816] app_queue.c: NO QUEUE_PRIO variable found. Using default. [Nov 4 19:50:35] DEBUG[1816] app_queue.c: queue: queue_support, options: tT, url: , announce: , expires: 1288898435, priority: 0 [Nov 4 19:50:35] DEBUG[1816] app_queue.c: Queue queue_support has no realtime members defined. No need for update [Nov 4 19:50:35] DEBUG[1816] app_queue.c: Queue 'queue_support' Join, Channel 'SIP/623-00000012', Position '1' [Nov 4 19:50:35] VERBOSE[1816] logger.c: -- Started music on hold, class 'default', on SIP/623-00000012 [Nov 4 19:50:35] DEBUG[1816] channel.c: Scheduling timer at 160 sample intervals [Nov 4 19:50:35] DEBUG[1816] channel.c: Prodding channel 'SIP/623-00000012' [Nov 4 19:50:35] DEBUG[1816] app_queue.c: There is 1 available member. [Nov 4 19:50:35] DEBUG[1816] app_queue.c: It's our turn (SIP/623-00000012). [Nov 4 19:50:35] DEBUG[1816] app_queue.c: SIP/623-00000012 is trying to call a queue member. [Nov 4 19:50:35] DEBUG[1816] app_queue.c: (Parallel) Trying 'SIP/671' with metric 0 [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Setting NAT on RTP to Off [Nov 4 19:50:35] DEBUG[1816] acl.c: ##### Testing 192.168.10.10 with 192.168.10.0 [Nov 4 19:50:35] DEBUG[1816] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** Our native formats are 0x80004 (ulaw|h263) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Nov 4 19:50:35] DEBUG[1816] frame.c: Could not find preferred codec - Going for the best codec [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: This channel will not be able to handle video. [Nov 4 19:50:35] DEBUG[1816] channel.c: Not copying variable SIPCALLID. [Nov 4 19:50:35] DEBUG[1816] channel.c: Not copying variable SIPUSERAGENT. [Nov 4 19:50:35] DEBUG[1816] channel.c: Not copying variable SIPDOMAIN. [Nov 4 19:50:35] DEBUG[1816] channel.c: Not copying variable SIPURI. [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Outgoing Call for 671 [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Call to peer '671' is 1 out of 2 [Nov 4 19:50:35] DEBUG[1816] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: ** Our capability: 0x8000e (gsm|ulaw|alaw|h263) Video flag: False [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: This call needs video offers, but there's no video support enabled! [Nov 4 19:50:35] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:35] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:35] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 6 (Ringing) [Nov 4 19:50:35] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '6' (Ringing) [Nov 4 19:50:35] VERBOSE[1816] logger.c: Audio is at 192.168.10.181 port 16104 [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding codec 0x2 (gsm) to SDP [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Done building SDP. Settling with this capability: 0x8000e (gsm|ulaw|alaw|h263) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 0: INVITE sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 (69) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport (65) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 2: From: "MR X" ;tag=as5c78aae1 (54) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 3: To: (60) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 4: Contact: (35) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 5: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 (56) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 9: Date: Thu, 04 Nov 2010 18:50:35 GMT (35) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO (72) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 11: Supported: replaces (19) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 13: Content-Length: 262 (19) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Header 14: (0) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: v=0 (3) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: o=root 1637 1637 IN IP4 192.168.10.181 (38) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: s=session (9) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: c=IN IP4 192.168.10.181 (23) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: t=0 0 (5) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: m=audio 16104 RTP/AVP 0 3 8 101 (31) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=ptime:20 (10) [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Line: a=sendrecv (10) [Nov 4 19:50:35] VERBOSE[1816] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:56602: INVITE sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport From: "MR X" ;tag=as5c78aae1 To: Contact: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 04 Nov 2010 18:50:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 1637 1637 IN IP4 192.168.10.181 s=session c=IN IP4 192.168.10.181 t=0 0 m=audio 16104 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 4 19:50:35] DEBUG[1816] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:35] DEBUG[1816] channel.c: Set channel SIP/623-00000012 to write format slin [Nov 4 19:50:35] DEBUG[1816] res_musiconhold.c: SIP/623-00000012 Opened file 1 '/var/lib/asterisk/moh/macroform-the_simplicity' [Nov 4 19:50:35] DEBUG[1816] rtp.c: Setting the marker bit due to a source update [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Setting framing from config on incoming call [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: ** Our capability: 0xc (ulaw|alaw) Video flag: True [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 4 19:50:35] VERBOSE[1816] logger.c: Audio is at 192.168.10.181 port 10606 [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding codec 0x4 (ulaw) to SDP [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 4 19:50:35] VERBOSE[1816] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: -- Done with adding codecs to SDP [Nov 4 19:50:35] DEBUG[1816] chan_sip.c: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Nov 4 19:50:35] VERBOSE[1816] logger.c: <--- Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7;received=192.168.10.163 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as29c61352 Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: Content-Type: application/sdp Content-Length: 239 v=0 o=root 1637 1637 IN IP4 192.168.10.181 s=session c=IN IP4 192.168.10.181 t=0 0 m=audio 10606 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Nov 4 19:50:35] DEBUG[1816] rtp.c: Ooh, format changed from unknown to ulaw [Nov 4 19:50:35] DEBUG[1816] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Nov 4 19:50:35] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.10:56602 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 Contact: To: ;tag=2972b36a From: "MR X";tag=as5c78aae1 Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 (70) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 2: Contact: (65) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 3: To: ;tag=2972b36a (73) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 4: From: "MR X";tag=as5c78aae1 (53) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 (56) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 7: User-Agent: 3CXPhone 4.0.10858.0 (32) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 8: Content-Length: 0 (17) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 9: (0) [Nov 4 19:50:35] VERBOSE[1644] logger.c: --- (9 headers 0 lines) --- [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: = Found Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag Our tag: as5c78aae1 [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: *** SIP TIMER: Cancelling retransmission #867 - INVITE (got response) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' Request 102: Found [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: SIP response 180 to standard invite [Nov 4 19:50:35] DEBUG[1644] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:35] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:35] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:35] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 6 (Ringing) [Nov 4 19:50:35] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '6' (Ringing) [Nov 4 19:50:35] VERBOSE[1816] logger.c: -- SIP/671-00000013 is ringing [Nov 4 19:50:35] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.154:5061 ---> NOTIFY sip:192.168.10.181:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-98cca4ff From: "gj linksys" ;tag=f78e9e178d81d8e8o0 To: Call-ID: 133d9d23-8e12054c@192.168.10.154 CSeq: 122 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA922-5.2.8 Content-Length: 0 <-------------> [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 0: NOTIFY sip:192.168.10.181:5060 SIP/2.0 (38) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-98cca4ff (60) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 2: From: "gj linksys" ;tag=f78e9e178d81d8e8o0 (66) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 3: To: (24) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 4: Call-ID: 133d9d23-8e12054c@192.168.10.154 (41) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 5: CSeq: 122 NOTIFY (16) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 6: Max-Forwards: 70 (16) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 7: Event: keep-alive (17) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 8: User-Agent: Linksys/SPA922-5.2.8 (32) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 9: Content-Length: 0 (17) [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Header 10: (0) [Nov 4 19:50:35] VERBOSE[1644] logger.c: --- (10 headers 0 lines) --- [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: = No match Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag 2972b36a Our tag: as5c78aae1 [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: = No match Their Call ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Their Tag 92BC5B1-A620971C Our tag: as29c61352 [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: = No match Their Call ID: d585d491-dc048da7-e478da@192.168.10.163 Their Tag 955EBB4D-7EA49688 Our tag: as0e622a65 [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: = No match Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:35] DEBUG[1644] acl.c: ##### Testing 192.168.10.154 with 192.168.10.0 [Nov 4 19:50:35] VERBOSE[1644] logger.c: Sending to 192.168.10.154 : 5061 (no NAT) [Nov 4 19:50:35] VERBOSE[1644] logger.c: <--- Transmitting (no NAT) to 192.168.10.154:5061 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.10.154:5061;branch=z9hG4bK-98cca4ff;received=192.168.10.154 From: "gj linksys" ;tag=f78e9e178d81d8e8o0 To: ;tag=as75079625 Call-ID: 133d9d23-8e12054c@192.168.10.154 CSeq: 122 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Nov 4 19:50:35] DEBUG[1644] chan_sip.c: Invalid SIP message - rejected , no callid, len 349 [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> CANCEL sip:500@192.168.10.181;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 From: "623" ;tag=92BC5B1-A620971C To: CSeq: 2 CANCEL Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="18a4d417ef54e647d9421c395f48abcb", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 0: CANCEL sip:500@192.168.10.181;user=phone SIP/2.0 (48) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 (61) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 2: From: "623" ;tag=92BC5B1-A620971C (57) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 3: To: (39) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 4: CSeq: 2 CANCEL (14) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 (50) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 6: Contact: (33) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 (54) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 8: Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="18a4d417ef54e647d9421c395f48abcb", algorithm=MD5 (179) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 10: Content-Length: 0 (17) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 11: (0) [Nov 4 19:50:39] VERBOSE[1644] logger.c: --- (11 headers 0 lines) --- [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = No match Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag 2972b36a Our tag: as5c78aae1 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = Found Their Call ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Their Tag 92BC5B1-A620971C Our tag: as29c61352 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL [Nov 4 19:50:39] VERBOSE[1644] logger.c: Sending to 192.168.10.163 : 5060 (no NAT) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Setting SIP_ALREADYGONE on dialog 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Updating call counter for incoming call [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Call from user '623' removed from call limit 2 [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7;received=192.168.10.163 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as29c61352 Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- Transmitting (no NAT) to 192.168.10.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7;received=192.168.10.163 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as29c61352 Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> [Nov 4 19:50:39] VERBOSE[1816] logger.c: -- Stopped music on hold on SIP/623-00000012 [Nov 4 19:50:39] DEBUG[1816] channel.c: Set channel SIP/623-00000012 to write format ulaw [Nov 4 19:50:39] DEBUG[1816] channel.c: Scheduling timer at 0 sample intervals [Nov 4 19:50:39] DEBUG[1816] app_queue.c: SIP/623-00000012: Nobody answered. [Nov 4 19:50:39] DEBUG[1816] channel.c: Hanging up channel 'SIP/671-00000013' [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Hangup call SIP/671-00000013, SIP callid 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181) [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: update_call_counter(671) - decrement call limit counter on hangup [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Call to peer '671' removed from call limit 2 [Nov 4 19:50:39] DEBUG[1816] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:39] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 19:50:39] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Hanging up channel in state Ringing (not UP) [Nov 4 19:50:39] VERBOSE[1816] logger.c: Scheduling destruction of SIP dialog '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' in 6976 ms (Method: INVITE) [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' Request 102: Found [Nov 4 19:50:39] VERBOSE[1816] logger.c: Reliably Transmitting (no NAT) to 192.168.10.10:56602: CANCEL sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport From: "MR X" ;tag=as5c78aae1 To: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #-1 [Nov 4 19:50:39] VERBOSE[1816] logger.c: Scheduling destruction of SIP dialog '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' in 6976 ms (Method: INVITE) [Nov 4 19:50:39] DEBUG[1816] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:39] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 19:50:39] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 19:50:39] DEBUG[1816] app_queue.c: Queue 'queue_support' Leave, Channel 'SIP/623-00000012' [Nov 4 19:50:39] DEBUG[1816] pbx.c: Spawn extension (do_support,s,4) exited non-zero on 'SIP/623-00000012' [Nov 4 19:50:39] VERBOSE[1816] logger.c: == Spawn extension (do_support, s, 4) exited non-zero on 'SIP/623-00000012' [Nov 4 19:50:39] DEBUG[1816] channel.c: Soft-Hanging up channel 'SIP/623-00000012' [Nov 4 19:50:39] DEBUG[1816] pbx.c: Launching 'Hangup' [Nov 4 19:50:39] VERBOSE[1816] logger.c: -- Executing [h@do_support:1] Hangup("SIP/623-00000012", "") in new stack [Nov 4 19:50:39] DEBUG[1816] pbx.c: Spawn extension (do_support,h,1) exited non-zero on 'SIP/623-00000012' [Nov 4 19:50:39] VERBOSE[1816] logger.c: == Spawn extension (do_support, h, 1) exited non-zero on 'SIP/623-00000012' [Nov 4 19:50:39] DEBUG[1816] channel.c: Hanging up channel 'SIP/623-00000012' [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Hangup call SIP/623-00000012, SIP callid 59ba5135-93a9d0ab-e25cb58e@192.168.10.163) [Nov 4 19:50:39] DEBUG[1816] chan_sip.c: Hanging up channel in state Ring (not UP) [Nov 4 19:50:39] DEBUG[1816] devicestate.c: Notification of state change to be queued on device/channel SIP/623 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 623 [Nov 4 19:50:39] DEBUG[1641] chan_sip.c: Checking device state for peer 623 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: Changing state for SIP/623 - state 1 (Not in use) [Nov 4 19:50:39] DEBUG[1662] app_queue.c: Device 'SIP/623' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> ACK sip:500@192.168.10.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 From: "623" ;tag=92BC5B1-A620971C To: ;tag=as29c61352 CSeq: 2 ACK Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="e8dcfe7a0818eff6a89a7bd90d2e8748", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 0: ACK sip:500@192.168.10.181 SIP/2.0 (34) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.163;branch=z9hG4bK5251a7aD99ED1C7 (61) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 2: From: "623" ;tag=92BC5B1-A620971C (57) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 3: To: ;tag=as29c61352 (54) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 4: CSeq: 2 ACK (11) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 (50) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 6: Contact: (33) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 (54) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 9: Proxy-Authorization: Digest username="623", realm="asterisk", nonce="146ab11d", uri="sip:500@192.168.10.181;user=phone", response="e8dcfe7a0818eff6a89a7bd90d2e8748", algorithm=MD5 (179) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 10: Max-Forwards: 70 (16) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 11: Content-Length: 0 (17) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 12: (0) [Nov 4 19:50:39] VERBOSE[1644] logger.c: --- (12 headers 0 lines) --- [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = No match Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag 2972b36a Our tag: as5c78aae1 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = Found Their Call ID: 59ba5135-93a9d0ab-e25cb58e@192.168.10.163 Their Tag 92BC5B1-A620971C Our tag: as29c61352 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #871 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Stopping retransmission on '59ba5135-93a9d0ab-e25cb58e@192.168.10.163' of Response 2: Match Found [Nov 4 19:50:39] VERBOSE[1644] logger.c: Really destroying SIP dialog '59ba5135-93a9d0ab-e25cb58e@192.168.10.163' Method: ACK [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.10:56602 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 Contact: To: ;tag=2972b36a From: "MR X";tag=as5c78aae1 Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 CANCEL User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 (70) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 2: Contact: (65) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 3: To: ;tag=2972b36a (73) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 4: From: "MR X";tag=as5c78aae1 (53) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 5: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 (56) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 6: CSeq: 102 CANCEL (16) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 7: User-Agent: 3CXPhone 4.0.10858.0 (32) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 8: Content-Length: 0 (17) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 9: (0) [Nov 4 19:50:39] VERBOSE[1644] logger.c: --- (9 headers 0 lines) --- [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = Found Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag 2972b36a Our tag: as5c78aae1 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Acked pending invite 102 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #873 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Stopping retransmission on '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' of Request 102: Match Found [Nov 4 19:50:39] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.10:56602 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 To: ;tag=2972b36a From: "MR X";tag=as5c78aae1 Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 INVITE User-Agent: 3CXPhone 4.0.10858.0 Content-Length: 0 <-------------> [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 0: SIP/2.0 487 Request Terminated (30) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport=5060 (70) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 2: To: ;tag=2972b36a (73) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 3: From: "MR X";tag=as5c78aae1 (53) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 4: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 (56) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 5: CSeq: 102 INVITE (16) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 6: User-Agent: 3CXPhone 4.0.10858.0 (32) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 7: Content-Length: 0 (17) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Header 8: (0) [Nov 4 19:50:39] VERBOSE[1644] logger.c: --- (8 headers 0 lines) --- [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: = Found Their Call ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 Their Tag 2972b36a Our tag: as5c78aae1 [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Stopping retransmission on '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' of Request 102: Match Found [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: SIP response 487 to standard invite [Nov 4 19:50:39] VERBOSE[1644] logger.c: Transmitting (no NAT) to 192.168.10.10:56602: ACK sip:671@192.168.10.10:56602;rinstance=6a21e91ae5778a42 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.181:5060;branch=z9hG4bK0a51eb4e;rport From: "MR X" ;tag=as5c78aae1 To: ;tag=2972b36a Contact: Call-ID: 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Updating call counter for outgoing call [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Call to peer '671' removed from call limit 2 [Nov 4 19:50:39] DEBUG[1644] devicestate.c: Notification of state change to be queued on device/channel SIP/671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: No provider found, checking channel drivers for SIP - 671 [Nov 4 19:50:39] DEBUG[1641] chan_sip.c: Checking device state for peer 671 [Nov 4 19:50:39] DEBUG[1641] devicestate.c: Changing state for SIP/671 - state 1 (Not in use) [Nov 4 19:50:39] DEBUG[1662] app_queue.c: Device 'SIP/671' changed to state '1' (Not in use) [Nov 4 19:50:39] DEBUG[1644] chan_sip.c: Setting SIP_ALREADYGONE on dialog 3c4512c56d84a61177c32efc3d33fc99@192.168.10.181 [Nov 4 19:50:39] VERBOSE[1644] logger.c: Really destroying SIP dialog '3c4512c56d84a61177c32efc3d33fc99@192.168.10.181' Method: INVITE [Nov 4 19:50:40] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.10:56602 ---> <-------------> [Nov 4 19:50:40] DEBUG[1644] chan_sip.c: Header 0: (0) [Nov 4 19:50:40] DEBUG[1644] chan_sip.c: Line: (0) [Nov 4 19:50:43] VERBOSE[1644] logger.c: <--- SIP read from 192.168.10.163:5060 ---> <-------------> [Nov 4 19:50:43] DEBUG[1644] chan_sip.c: Header 0: (0) [Nov 4 19:50:43] DEBUG[1644] chan_sip.c: Line: (0) [Nov 4 19:50:50] DEBUG[1644] chan_sip.c: = No match Their Call ID: d585d491-dc048da7-e478da@192.168.10.163 Their Tag 955EBB4D-7EA49688 Our tag: as0e622a65 [Nov 4 19:50:50] DEBUG[1644] chan_sip.c: = No match Their Call ID: ODBkNjNkNTE3ZjFhNDMwMmM1NGUwNmFlZDBhMDgyYTU. Their Tag 932d992d Our tag: as18545d7b [Nov 4 19:50:50] DEBUG[1644] acl.c: ##### Testing 192.168.10.154 with 192.168.10.0 [Nov 4 19:50:50] DEBUG[1644] chan_sip.c: Invalid SIP message - rejected , no callid, len 349