[Jan 17 22:59:13] NOTICE[4935] loader.c: 1 modules will be loaded. [Jan 17 22:59:13] NOTICE[4935] cdr.c: CDR logging disabled, data will be lost. [Jan 17 22:59:13] NOTICE[4935] loader.c: 201 modules will be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Error loading module 'chan_oss.so': libX11.so.6: cannot open shared object file: No such file or directory [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_calendar.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_calendar.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_calendar.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_indications.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_indications.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_indications.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_smdi.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_smdi.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_smdi.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_pgsql.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_pgsql.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_pgsql.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Error loading module 'app_dahdiscan.so': /usr/lib/asterisk/modules/app_dahdiscan.so: undefined symbol: ast_channel_walk_locked [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'app_dahdiscan.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_curl.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_curl.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_curl.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Error loading module 'res_calendar_caldav.so': /usr/lib/asterisk/modules/res_calendar_caldav.so: undefined symbol: ast_calendar_event_container_alloc [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_calendar_caldav.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Error loading module 'chan_oss.so': libX11.so.6: cannot open shared object file: No such file or directory [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'chan_oss.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_custom.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_custom.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_custom.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_csv.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_csv.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_csv.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_ldap.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_ldap.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_config_ldap.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_syslog.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_syslog.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_syslog.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_custom.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_custom.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_custom.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Error loading module 'res_calendar_icalendar.so': /usr/lib/asterisk/modules/res_calendar_icalendar.so: undefined symbol: ast_calendar_event_container_alloc [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'res_calendar_icalendar.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_manager.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_manager.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cel_manager.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_manager.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_manager.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_manager.so' could not be loaded. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_pgsql.so' was not compiled with the same compile-time options as this version of Asterisk. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_pgsql.so' will not be initialized as it may cause instability. [Jan 17 22:59:13] WARNING[4935] loader.c: Module 'cdr_pgsql.so' could not be loaded. [Jan 17 22:59:13] NOTICE[4935] chan_skinny.c: Configuring skinny from skinny.conf [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Detected alarm on channel 1: Red Alarm [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23. [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35. [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [Jan 17 22:59:13] WARNING[4935] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Jan 17 22:59:13] VERBOSE[4935] chan_sip.c: SIP channel loading... [Jan 17 22:59:13] NOTICE[4953] chan_sip.c: Peer '*11' is now Reachable. (4ms / 2000ms) [Jan 17 22:59:13] NOTICE[4953] chan_sip.c: Peer '*10' is now Reachable. (104ms / 2000ms) [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: Starting AEL load process. [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Jan 17 22:59:13] NOTICE[4935] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jan 17 22:59:13] VERBOSE[4935] res_clialiases.c: == Aliased CLI command 'hangup request' to 'channel request hangup' [Jan 17 22:59:13] VERBOSE[4935] res_clialiases.c: == Aliased CLI command 'originate' to 'channel originate' [Jan 17 22:59:13] VERBOSE[4935] res_clialiases.c: == Aliased CLI command 'help' to 'core show help' [Jan 17 22:59:13] VERBOSE[4935] res_clialiases.c: == Aliased CLI command 'pri intense debug span' to 'pri set debug 2 span' [Jan 17 22:59:13] VERBOSE[4935] res_clialiases.c: == Aliased CLI command 'reload' to 'module reload' [Jan 17 22:59:13] NOTICE[4953] chan_sip.c: Received SIP subscribe for peer without mailbox: *11 [Jan 17 22:59:38] VERBOSE[4980] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 17 22:59:38] DEBUG[4980] config.c: Parsing /etc/asterisk/logger.conf [Jan 17 22:59:38] VERBOSE[4980] config.c: == Found [Jan 17 22:59:38] VERBOSE[4980] logger.c: Asterisk Queue Logger restarted [Jan 17 22:59:40] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> <-------------> [Jan 17 22:59:40] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 22:59:53] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> <-------------> [Jan 17 22:59:53] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 23:00:10] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> <-------------> [Jan 17 23:00:10] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Allocating new SIP dialog for 16861cb5264d01f95c161ff437b9aaa7@192.168.15.21:0 - OPTIONS (No RTP) [Jan 17 23:00:13] DEBUG[4953] acl.c: For destination '192.168.15.5', our source address is '192.168.15.21'. [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Initializing initreq for method OPTIONS - callid 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 0 [ 69]: OPTIONS sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK75b45a19 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 3 [ 58]: From: "CallBox" ;tag=as72dae529 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 4 [ 59]: To: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 5 [ 41]: Contact: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 6 [ 60]: Call-ID: 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 9 [ 35]: Date: Mon, 17 Jan 2011 22:00:13 GMT [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.15.5:58988: OPTIONS sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK75b45a19 Max-Forwards: 70 From: "CallBox" ;tag=as72dae529 To: Contact: Call-ID: 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 17 Jan 2011 22:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #15 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.15.5:58988 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK75b45a19 Contact: To: ;tag=c65fbc7c From: "CallBox";tag=as72dae529 Call-ID: 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 0 <-------------> [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK75b45a19 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 2 [ 33]: Contact: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 3 [ 72]: To: ;tag=c65fbc7c [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 4 [ 57]: From: "CallBox";tag=as72dae529 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 5 [ 60]: Call-ID: 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 7 [ 23]: Accept: application/sdp [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 10 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: --- (12 headers 0 lines) --- [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: = Looking for Call ID: 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 (Checking To) --From tag as72dae529 --To-tag c65fbc7c [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Stopping retransmission on '20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060' of Request 102: Match Found [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Destroying SIP dialog 20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: Really destroying SIP dialog '20ef92ab69491fa63bee752d246270a0@192.168.15.21:5060' Method: OPTIONS [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Allocating new SIP dialog for 26552a9b2eccd0285f68fe4f15222382@192.168.15.21:0 - OPTIONS (No RTP) [Jan 17 23:00:13] DEBUG[4953] acl.c: For destination '192.168.15.5', our source address is '192.168.15.21'. [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Initializing initreq for method OPTIONS - callid 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 0 [ 69]: OPTIONS sip:*10@192.168.15.5:13664;rinstance=c7e1789db03f8822 SIP/2.0 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK0d3d2a48 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 3 [ 58]: From: "CallBox" ;tag=as620b0d2f [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 4 [ 59]: To: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 5 [ 41]: Contact: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 6 [ 60]: Call-ID: 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 9 [ 35]: Date: Mon, 17 Jan 2011 22:00:13 GMT [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.15.5:13664: OPTIONS sip:*10@192.168.15.5:13664;rinstance=c7e1789db03f8822 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK0d3d2a48 Max-Forwards: 70 From: "CallBox" ;tag=as620b0d2f To: Contact: Call-ID: 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 17 Jan 2011 22:00:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK0d3d2a48 Contact: To: ;tag=8c4b8408 From: "CallBox";tag=as620b0d2f Call-ID: 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 0 <-------------> [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK0d3d2a48 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 2 [ 33]: Contact: [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 3 [ 72]: To: ;tag=8c4b8408 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 4 [ 57]: From: "CallBox";tag=as620b0d2f [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 5 [ 60]: Call-ID: 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 6 [ 17]: CSeq: 102 OPTIONS [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 7 [ 23]: Accept: application/sdp [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 8 [ 19]: Accept-Language: en [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: --- (12 headers 0 lines) --- [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: = Looking for Call ID: 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 (Checking To) --From tag as620b0d2f --To-tag 8c4b8408 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Stopping retransmission on '1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060' of Request 102: Match Found [Jan 17 23:00:13] DEBUG[4953] chan_sip.c: Destroying SIP dialog 1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060 [Jan 17 23:00:13] VERBOSE[4953] chan_sip.c: Really destroying SIP dialog '1ce4a551078feaad0e4a96f531f0012a@192.168.15.21:5060' Method: OPTIONS [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> INVITE sip:*11@192.168.15.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-ae239629ae884d47-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Thinkro *10 (Softphone)" From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Content-Length: 168 v=0 o=- 6 2 IN IP4 192.168.15.5 s=CallBox c=IN IP4 192.168.15.5 t=0 0 m=audio 61580 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 0 [ 36]: INVITE sip:*11@192.168.15.21 SIP/2.0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-ae239629ae884d47-1---d8754z-;rport [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 4 [ 52]: To: "Thinkro *10 (Softphone)" [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 5 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 6 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 7 [ 14]: CSeq: 1 INVITE [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 11 [ 19]: Content-Length: 168 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 12 [ 0]: [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 1 [ 27]: o=- 6 2 IN IP4 192.168.15.5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 2 [ 9]: s=CallBox [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 5 [ 27]: m=audio 61580 RTP/AVP 8 101 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: --- (12 headers 9 lines) --- [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag [Jan 17 23:00:14] DEBUG[4953] acl.c: For destination '192.168.15.5', our source address is '192.168.15.21'. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.21:5060 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Allocating new SIP dialog for ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. - INVITE (No RTP) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 17 23:00:14] DEBUG[4953] netsock2.c: Splitting '192.168.15.5:13664' gives... [Jan 17 23:00:14] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '13664'. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Sending to 192.168.15.5:13664 (NAT) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Initializing initreq for method INVITE - callid ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Using INVITE request as basis request - ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Found peer '*10' for '*10' from 192.168.15.5:13664 [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.15.5:13664 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-ae239629ae884d47-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)";tag=as1b2cf78b Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="063f92ab" Content-Length: 0 <------------> [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #21 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Scheduling destruction of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' in 6592 ms (Method: INVITE) [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> ACK sip:*11@192.168.15.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-ae239629ae884d47-1---d8754z-;rport To: "Thinkro *10 (Softphone)";tag=as1b2cf78b From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 1 ACK Content-Length: 0 <-------------> [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 0 [ 33]: ACK sip:*11@192.168.15.21 SIP/2.0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-ae239629ae884d47-1---d8754z-;rport [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 2 [ 67]: To: "Thinkro *10 (Softphone)";tag=as1b2cf78b [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 3 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 4 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: --- (7 headers 0 lines) --- [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag as1b2cf78b [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #21 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Stopping retransmission on 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' of Response 1: Match Found [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> INVITE sip:*11@192.168.15.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Thinkro *10 (Softphone)" From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 Content-Length: 168 v=0 o=- 6 2 IN IP4 192.168.15.5 s=CallBox c=IN IP4 192.168.15.5 t=0 0 m=audio 61580 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 0 [ 36]: INVITE sip:*11@192.168.15.21 SIP/2.0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;rport [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 4 [ 52]: To: "Thinkro *10 (Softphone)" [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 5 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 6 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 7 [ 14]: CSeq: 2 INVITE [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 10 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 11 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 12 [ 19]: Content-Length: 168 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Header 13 [ 0]: [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 1 [ 27]: o=- 6 2 IN IP4 192.168.15.5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 2 [ 9]: s=CallBox [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 5 [ 27]: m=audio 61580 RTP/AVP 8 101 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 6 [ 15]: a=fmtp:101 0-15 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: --- (13 headers 9 lines) --- [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag [Jan 17 23:00:14] DEBUG[4953] netsock2.c: Splitting '192.168.15.21' gives... [Jan 17 23:00:14] DEBUG[4953] netsock2.c: ...host '192.168.15.21' and port '(null)'. [Jan 17 23:00:14] DEBUG[4953] netsock2.c: Splitting '192.168.15.21' gives... [Jan 17 23:00:14] DEBUG[4953] netsock2.c: ...host '192.168.15.21' and port '(null)'. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 17 23:00:14] DEBUG[4953] netsock2.c: Splitting '192.168.15.5:13664' gives... [Jan 17 23:00:14] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '13664'. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Sending to 192.168.15.5:13664 (no NAT) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Initializing initreq for method INVITE - callid ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Using INVITE request as basis request - ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Found peer '*10' for '*10' from 192.168.15.5:13664 [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x87808c0' [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Allocated port 14684 for RTP instance '0x87808c0' [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: RTP instance '0x87808c0' is setup and ready to go [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x8784b50' [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Allocated port 10932 for RTP instance '0x8784b50' [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: RTP instance '0x8784b50' is setup and ready to go [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x8784b50' [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x87808c0' [Jan 17 23:00:14] VERBOSE[4953] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Setting NAT on RTP to On [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Setting NAT on VRTP to On [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing session-level SDP o=- 6 2 IN IP4 192.168.15.5... UNSUPPORTED. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing session-level SDP s=CallBox... UNSUPPORTED. [Jan 17 23:00:14] DEBUG[4953] netsock2.c: Splitting '192.168.15.5' gives... [Jan 17 23:00:14] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '(null)'. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.5... OK. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Found RTP audio format 8 [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Setting payload 8 based on m type on 0xb374e6e0 [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Found RTP audio format 101 [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Setting payload 101 based on m type on 0xb374e6e0 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Incorporating payload 8 on 0xb374e6e0 [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Incorporating payload 101 on 0xb374e6e0 [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x87808c0' [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Peer audio RTP is at port 192.168.15.5:61580 [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Copying payload 8 from 0xb374e6e0 to 0x8780a6c [Jan 17 23:00:14] DEBUG[4953] rtp_engine.c: Copying payload 101 from 0xb374e6e0 to 0x8780a6c [Jan 17 23:00:14] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8784b50' [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Peer doesn't provide video [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Checking SIP call limits for device *10 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Updating call counter for incoming call [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: Looking for *11 in local (domain 192.168.15.21) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: *** Our capabilities are 0x80008 (alaw|h263) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jan 17 23:00:14] DEBUG[4953] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 17 23:00:14] DEBUG[4953] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: build_route: Contact hop: [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: list_route: hop: [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: SIP/*10-00000000: New call is still down.... Trying... [Jan 17 23:00:14] VERBOSE[4953] chan_sip.c: <--- Transmitting (no NAT) to 192.168.15.5:13664 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)" Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 17 23:00:14] DEBUG[4953] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:14] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 23:00:14] DEBUG[4941] chan_sip.c: Checking device state for peer *10 [Jan 17 23:00:14] DEBUG[4941] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 23:00:14] DEBUG[4941] devicestate.c: device 'SIP/*10' state '1' [Jan 17 23:00:14] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:14.930759','CHAN_START','','XLite','*10','','','','*11','local','SIP/*10-00000000','','','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:14] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:14] DEBUG[4986] pbx.c: Result of 'EXTEN' is '*11' [Jan 17 23:00:14] DEBUG[4986] pbx.c: Launching 'Dial' [Jan 17 23:00:14] VERBOSE[4986] pbx.c: -- Executing [*11@local:1] Dial("SIP/*10-00000000", "SIP/*11,60,tTkK") in new stack [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Allocating new SIP dialog for 13fc3e9d03a0d061104c307161220353@192.168.15.21:0 - INVITE (No RTP) [Jan 17 23:00:14] DEBUG[4986] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x877c8b8' [Jan 17 23:00:14] DEBUG[4986] res_rtp_asterisk.c: Allocated port 10370 for RTP instance '0x877c8b8' [Jan 17 23:00:14] DEBUG[4986] rtp_engine.c: RTP instance '0x877c8b8' is setup and ready to go [Jan 17 23:00:14] DEBUG[4986] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x87765f8' [Jan 17 23:00:14] DEBUG[4986] res_rtp_asterisk.c: Allocated port 10628 for RTP instance '0x87765f8' [Jan 17 23:00:14] DEBUG[4986] rtp_engine.c: RTP instance '0x87765f8' is setup and ready to go [Jan 17 23:00:14] DEBUG[4986] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x87765f8' [Jan 17 23:00:14] DEBUG[4986] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x877c8b8' [Jan 17 23:00:14] VERBOSE[4986] netsock2.c: == Using SIP RTP CoS mark 5 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Setting NAT on RTP to On [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Setting NAT on VRTP to On [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 17 23:00:14] DEBUG[4986] acl.c: For destination '192.168.15.5', our source address is '192.168.15.21'. [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.15.21:5060 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** Our capabilities are 0x80008 (alaw|h263) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jan 17 23:00:14] DEBUG[4986] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 17 23:00:14] DEBUG[4986] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 17 23:00:14] DEBUG[4986] rtp_engine.c: Seeded SDP of 'SIP/*11-00000001' with that of 'SIP/*10-00000000' [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable DIALEDTIME. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable ANSWEREDTIME. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable DIALEDPEERNAME. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable DIALSTATUS. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable SIPCALLID. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable SIPDOMAIN. [Jan 17 23:00:14] DEBUG[4986] channel.c: Not copying variable SIPURI. [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Outgoing Call for *11 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Updating call counter for outgoing call [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: This call needs video offers! [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: ** Our capability: 0x80008 (alaw|h263) Video flag: False Text flag: False [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jan 17 23:00:14] VERBOSE[4986] chan_sip.c: Audio is at 5060 [Jan 17 23:00:14] VERBOSE[4986] chan_sip.c: Video is at 192.168.15.21:5060 [Jan 17 23:00:14] VERBOSE[4986] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 17 23:00:14] VERBOSE[4986] chan_sip.c: Adding video codec 0x80000 (h263) to SDP [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: -- Done with adding codecs to SDP [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Done building SDP. Settling with this capability: 0x80008 (alaw|h263) [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Initializing initreq for method INVITE - callid 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 0 [ 68]: INVITE sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 3 [ 52]: From: "XLite" ;tag=as50289726 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 4 [ 59]: To: [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 5 [ 37]: Contact: [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 6 [ 60]: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 8 [ 24]: User-Agent: Asterisk PBX [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 9 [ 35]: Date: Mon, 17 Jan 2011 22:00:14 GMT [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 17 23:00:14] VERBOSE[4986] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.15.5:58988: INVITE sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 Max-Forwards: 70 From: "XLite" ;tag=as50289726 To: Contact: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 17 Jan 2011 22:00:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 244 v=0 o=root 468764633 468764633 IN IP4 192.168.15.21 s=Asterisk PBX c=IN IP4 192.168.15.21 b=CT:384 t=0 0 m=audio 10370 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv m=video 10628 RTP/AVP 34 a=rtpmap:34 H263/90000 a=sendrecv --- [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #24 [Jan 17 23:00:14] DEBUG[4986] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.15.5:58988 [Jan 17 23:00:14] VERBOSE[4986] app_dial.c: -- Called *11 [Jan 17 23:00:14] DEBUG[4978] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 23:00:14] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:14.933298','APP_START','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','Dial','SIP/*11,60,tTkK','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:14] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:14] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:14.934663','CHAN_START','','','','','','','s','local','SIP/*11-00000001','','','','','1295301614.1','1295301614.0',3,'','')] [Jan 17 23:00:14] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:15] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 To: From: "XLite" ;tag=as50289726 Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 102 INVITE Content-Length: 0 <-------------> [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 2 [ 59]: To: [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 3 [ 52]: From: "XLite" ;tag=as50289726 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 4 [ 60]: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 6 [ 17]: Content-Length: 0 [Jan 17 23:00:15] VERBOSE[4953] chan_sip.c: --- (7 headers 0 lines) --- [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: = Looking for Call ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 (Checking To) --From tag as50289726 --To-tag [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: *** SIP TIMER: Cancelling retransmission #24 - INVITE (got response) [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Request 102: Found [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: SIP response 100 to standard invite [Jan 17 23:00:15] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 Contact: To: ;tag=6481f832 From: "XLite";tag=as50289726 Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 102 INVITE User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 0 <-------------> [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 2 [ 64]: Contact: [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 3 [ 72]: To: ;tag=6481f832 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 4 [ 51]: From: "XLite";tag=as50289726 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 5 [ 60]: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 7 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 17 23:00:15] VERBOSE[4953] chan_sip.c: --- (9 headers 0 lines) --- [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: = Looking for Call ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 (Checking To) --From tag as50289726 --To-tag 6481f832 [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Request 102: Found [Jan 17 23:00:15] DEBUG[4953] chan_sip.c: SIP response 180 to standard invite [Jan 17 23:00:15] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 23:00:15] DEBUG[4941] chan_sip.c: Checking device state for peer *11 [Jan 17 23:00:15] DEBUG[4941] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 23:00:15] DEBUG[4941] devicestate.c: device 'SIP/*11' state '1' [Jan 17 23:00:15] DEBUG[4978] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 23:00:15] VERBOSE[4986] app_dial.c: -- SIP/*11-00000001 is ringing [Jan 17 23:00:15] DEBUG[4986] rtp_engine.c: Setting early bridge SDP of 'SIP/*10-00000000' with that of 'SIP/*11-00000001' [Jan 17 23:00:15] VERBOSE[4986] chan_sip.c: <--- Transmitting (no NAT) to 192.168.15.5:13664 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)";tag=as70772d8f Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 17 23:00:15] DEBUG[4986] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:23] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> <-------------> [Jan 17 23:00:23] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.15.5:30463 [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: RTCP NAT: Got RTCP from other end. Now sending to address 192.168.15.5:54567 [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 Contact: To: ;tag=6481f832 From: "XLite";tag=as50289726 Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 213 v=0 o=- 1 2 IN IP4 192.168.15.5 s=CounterPath X-Lite 4.0 c=IN IP4 192.168.15.5 t=0 0 m=audio 30462 RTP/AVP 8 a=sendrecv m=video 54566 RTP/AVP 34 a=inactive a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=2;CIF=3 <-------------> [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK712b3002 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 2 [ 64]: Contact: [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 3 [ 72]: To: ;tag=6481f832 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 4 [ 51]: From: "XLite";tag=as50289726 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 5 [ 60]: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 6 [ 16]: CSeq: 102 INVITE [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 7 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 9 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 10 [ 19]: Content-Length: 213 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 11 [ 0]: [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 0 [ 3]: v=0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 1 [ 27]: o=- 1 2 IN IP4 192.168.15.5 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 2 [ 24]: s=CounterPath X-Lite 4.0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.15.5 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 5 [ 23]: m=audio 30462 RTP/AVP 8 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 6 [ 10]: a=sendrecv [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 7 [ 24]: m=video 54566 RTP/AVP 34 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 8 [ 10]: a=inactive [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 9 [ 22]: a=rtpmap:34 H263/90000 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Body 10 [ 22]: a=fmtp:34 QCIF=2;CIF=3 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: --- (11 headers 11 lines) --- [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: = Looking for Call ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 (Checking To) --From tag as50289726 --To-tag 6481f832 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Acked pending invite 102 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Stopping retransmission on '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' of Request 102: Match Found [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: SIP response 200 to standard invite [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing session-level SDP o=- 1 2 IN IP4 192.168.15.5... UNSUPPORTED. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing session-level SDP s=CounterPath X-Lite 4.0... UNSUPPORTED. [Jan 17 23:00:24] DEBUG[4953] netsock2.c: Splitting '192.168.15.5' gives... [Jan 17 23:00:24] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '(null)'. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.15.5... OK. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Found RTP audio format 8 [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Setting payload 8 based on m type on 0xb374ed30 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Found RTP video format 34 [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Setting payload 34 based on m type on 0xb374e0b0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing media-level (video) SDP a=inactive... OK. [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Found video description format H263 for ID 34 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing media-level (video) SDP a=rtpmap:34 H263/90000... OK. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Processing media-level (video) SDP a=fmtp:34 QCIF=2;CIF=3... UNSUPPORTED. [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Incorporating payload 8 on 0xb374ed30 [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Incorporating payload 34 on 0xb374e0b0 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Capabilities: us - 0x80008 (alaw|h263), peer - audio=0x8 (alaw)/video=0x80000 (h263)/text=0x0 (nothing), combined - 0x80008 (alaw|h263) [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jan 17 23:00:24] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x877c8b8' [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Peer audio RTP is at port 192.168.15.5:30462 [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Copying payload 8 from 0xb374ed30 to 0x877ca64 [Jan 17 23:00:24] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x87765f8' [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Peer video RTP is at port 192.168.15.5:54566 [Jan 17 23:00:24] DEBUG[4953] rtp_engine.c: Copying payload 34 from 0xb374e0b0 to 0x87767a4 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: We're settling with these formats: 0x80008 (alaw|h263) [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: We have an owner, now see if we need to change this call [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Updating call counter for outgoing call [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: build_route: Contact hop: [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: list_route: hop: [Jan 17 23:00:24] DEBUG[4953] netsock2.c: Splitting '192.168.15.5:58988' gives... [Jan 17 23:00:24] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '58988'. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Strict routing enforced for session 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 17 23:00:24] DEBUG[4953] netsock2.c: Splitting '192.168.15.5:58988' gives... [Jan 17 23:00:24] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '58988'. [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: set_destination: set destination to 192.168.15.5:58988 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Transmitting (no NAT) to 192.168.15.5:58988: ACK sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK6e35faca Max-Forwards: 70 From: "XLite" ;tag=as50289726 To: ;tag=6481f832 Contact: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Trying to put 'ACK sip:*11' onto UDP socket destined for 192.168.15.5:58988 [Jan 17 23:00:24] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 23:00:24] DEBUG[4941] chan_sip.c: Checking device state for peer *11 [Jan 17 23:00:24] DEBUG[4941] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 23:00:24] DEBUG[4941] devicestate.c: device 'SIP/*11' state '1' [Jan 17 23:00:24] VERBOSE[4986] app_dial.c: -- SIP/*11-00000001 answered SIP/*10-00000000 [Jan 17 23:00:24] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 23:00:24] DEBUG[4941] chan_sip.c: Checking device state for peer *10 [Jan 17 23:00:24] DEBUG[4941] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 23:00:24] DEBUG[4941] devicestate.c: device 'SIP/*10' state '1' [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: SIP answering channel: SIP/*10-00000000 [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: Setting framing from config on incoming call [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 17 23:00:24] VERBOSE[4986] chan_sip.c: Audio is at 5060 [Jan 17 23:00:24] VERBOSE[4986] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: -- Done with adding codecs to SDP [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jan 17 23:00:24] DEBUG[4978] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 23:00:24] VERBOSE[4986] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.15.5:13664 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)";tag=as70772d8f Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 172 v=0 o=root 384334276 384334276 IN IP4 192.168.15.21 s=Asterisk PBX c=IN IP4 192.168.15.21 t=0 0 m=audio 14684 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv <------------> [Jan 17 23:00:24] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:24.461499','ANSWER','','','*11','','','','*11','local','SIP/*11-00000001','AppDial','(Outgoing Line)','','','1295301614.1','1295301614.0',3,'','')] [Jan 17 23:00:24] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #27 [Jan 17 23:00:24] DEBUG[4986] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:24] DEBUG[4986] features.c: bridge answer set, chan answer set [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 17 23:00:24] VERBOSE[4986] rtp_engine.c: -- Locally bridging SIP/*10-00000000 and SIP/*11-00000001 [Jan 17 23:00:24] DEBUG[4978] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 23:00:24] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:24.462690','ANSWER','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','Dial','SIP/*11,60,tTkK','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:24] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:24] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:24.462805','BRIDGE_START','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','Dial','SIP/*11,60,tTkK','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:24] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:24] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: SIP TIMER: Rescheduling retransmission #27 (1) SIP/2.0 - 1 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 206 ms (t1 103 ms (Retrans id #27)) [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.15.5:13664: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-f768fe6d4e683f7e-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)";tag=as70772d8f Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 172 v=0 o=root 384334276 384334276 IN IP4 192.168.15.21 s=Asterisk PBX c=IN IP4 192.168.15.21 t=0 0 m=audio 14684 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: INVITE [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> ACK sip:*11@192.168.15.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-02b142538c2a8b5a-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Thinkro *10 (Softphone)";tag=as70772d8f From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 ACK User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 Content-Length: 0 <-------------> [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 0 [ 38]: ACK sip:*11@192.168.15.21:5060 SIP/2.0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-02b142538c2a8b5a-1---d8754z-;rport [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 4 [ 67]: To: "Thinkro *10 (Softphone)";tag=as70772d8f [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 5 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 6 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 9 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: --- (11 headers 0 lines) --- [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag as70772d8f [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Stopping retransmission on 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' of Response 2: Match Found [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> ACK sip:*11@192.168.15.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-02b142538c2a8b5a-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Thinkro *10 (Softphone)";tag=as70772d8f From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 2 ACK User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 Content-Length: 0 <-------------> [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 0 [ 38]: ACK sip:*11@192.168.15.21:5060 SIP/2.0 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-02b142538c2a8b5a-1---d8754z-;rport [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 4 [ 67]: To: "Thinkro *10 (Softphone)";tag=as70772d8f [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 5 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 6 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 7 [ 11]: CSeq: 2 ACK [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 9 [156]: Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21",response="3eb85b4d8c8e8e04ef1d0a3dd8624366",algorithm=MD5 [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 17 23:00:24] VERBOSE[4953] chan_sip.c: --- (11 headers 0 lines) --- [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag as70772d8f [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:24] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:24] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:24] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:24] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:25] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:25] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:26] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:26] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:27] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:27] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 152 bytes [Jan 17 23:00:27] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:27] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:27] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:28] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:28] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:29] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:29] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:29] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:29] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:30] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:30] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:30] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:30] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:30] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:31] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:31] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:32] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:32] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:33] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:33] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:33] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:33] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:33] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:34] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:34] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:35] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:35] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:35] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:36] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:36] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:36] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:36] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:36] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:37] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:37] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:38] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:38] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:39] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:39] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:39] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:39] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:39] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:40] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> <-------------> [Jan 17 23:00:40] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 23:00:40] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:40] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:41] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:41] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:42] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:42] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:42] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:42] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:42] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:43] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:43] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:44] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:44] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:44] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:44] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:45] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:45] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:45] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:45] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:45] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:45] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:46] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:46] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:47] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:47] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:48] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:48] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:48] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:48] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:48] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:49] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:49] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:50] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:50] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:51] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:51] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:51] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:51] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:51] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:52] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:52] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:53] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> <-------------> [Jan 17 23:00:53] DEBUG[4953] chan_sip.c: Header 0 [ 0]: [Jan 17 23:00:53] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:53] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:54] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:54] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:54] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 184 bytes [Jan 17 23:00:54] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 156 bytes [Jan 17 23:00:54] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 132 bytes [Jan 17 23:00:54] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 176 bytes [Jan 17 23:00:54] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:54] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: ACK [Jan 17 23:00:55] NOTICE[4986] res_rtp_asterisk.c: Unknown RTP codec 126 received from '192.168.15.5:54566' [Jan 17 23:00:55] DEBUG[4986] res_rtp_asterisk.c: Got RTCP report of 160 bytes [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:13664 ---> BYE sip:*11@192.168.15.21:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-8af0353055dca02b-1---d8754z-;rport Max-Forwards: 70 Contact: To: "Thinkro *10 (Softphone)";tag=as70772d8f From: "XLite";tag=aeea9751 Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 3 BYE User-Agent: X-Lite release 1014k stamp 47051 Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21:5060",response="9a25afbdbe99ecdbc9d206991a75e21a",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 <-------------> [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 0 [ 38]: BYE sip:*11@192.168.15.21:5060 SIP/2.0 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 1 [ 92]: Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-8af0353055dca02b-1---d8754z-;rport [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 3 [ 37]: Contact: [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 4 [ 67]: To: "Thinkro *10 (Softphone)";tag=as70772d8f [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 5 [ 49]: From: "XLite";tag=aeea9751 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 6 [ 53]: Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 7 [ 11]: CSeq: 3 BYE [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 8 [ 44]: User-Agent: X-Lite release 1014k stamp 47051 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 9 [161]: Authorization: Digest username="*10",realm="asterisk",nonce="063f92ab",uri="sip:*11@192.168.15.21:5060",response="9a25afbdbe99ecdbc9d206991a75e21a",algorithm=MD5 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 10 [ 38]: Reason: SIP;description="User Hung Up" [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: --- (12 headers 0 lines) --- [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: = Looking for Call ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. (Checking From) --From tag aeea9751 --To-tag as70772d8f [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Initializing initreq for method BYE - callid ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:55] DEBUG[4953] netsock2.c: Splitting '192.168.15.5:13664' gives... [Jan 17 23:00:55] DEBUG[4953] netsock2.c: ...host '192.168.15.5' and port '13664'. [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: Sending to 192.168.15.5:13664 (no NAT) [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Setting SIP_ALREADYGONE on dialog ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:55] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x87808c0' [Jan 17 23:00:55] DEBUG[4953] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8784b50' [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: Scheduling destruction of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' in 6592 ms (Method: BYE) [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Received bye, issuing owner hangup [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: <--- Transmitting (no NAT) to 192.168.15.5:13664 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.5:13664;branch=z9hG4bK-d8754z-8af0353055dca02b-1---d8754z-;received=192.168.15.5;rport=13664 From: "XLite";tag=aeea9751 To: "Thinkro *10 (Softphone)";tag=as70772d8f Call-ID: ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. CSeq: 3 BYE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.15.5:13664 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: BYE [Jan 17 23:00:55] DEBUG[4986] rtp_engine.c: rtp-engine-local-bridge: Ooh, got a hangup [Jan 17 23:00:55] DEBUG[4986] channel.c: Returning from native bridge, channels: SIP/*10-00000000, SIP/*11-00000001 [Jan 17 23:00:55] DEBUG[4986] pbx.c: Launching 'Hangup' [Jan 17 23:00:55] VERBOSE[4986] pbx.c: -- Executing [h@local:1] Hangup("SIP/*10-00000000", "") in new stack [Jan 17 23:00:55] DEBUG[4986] cdr.c: Dropping CDR ! [Jan 17 23:00:55] DEBUG[4986] channel.c: Hanging up channel 'SIP/*11-00000001' [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.614970','BRIDGE_END','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','Dial','SIP/*11,60,tTkK','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4986] chan_sip.c: Hangup call SIP/*11-00000001, SIP callid 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] DEBUG[4986] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x877c8b8' [Jan 17 23:00:55] DEBUG[4986] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x87765f8' [Jan 17 23:00:55] VERBOSE[4986] chan_sip.c: Scheduling destruction of SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' in 6400 ms (Method: INVITE) [Jan 17 23:00:55] DEBUG[4986] chan_sip.c: Strict routing enforced for session 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:55] VERBOSE[4986] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 17 23:00:55] DEBUG[4986] netsock2.c: Splitting '192.168.15.5:58988' gives... [Jan 17 23:00:55] DEBUG[4986] netsock2.c: ...host '192.168.15.5' and port '58988'. [Jan 17 23:00:55] VERBOSE[4986] chan_sip.c: set_destination: set destination to 192.168.15.5:58988 [Jan 17 23:00:55] VERBOSE[4986] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.15.5:58988: BYE sip:*11@192.168.15.5:58988;rinstance=0e9271158687d576 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK6a771751 Max-Forwards: 70 From: "XLite" ;tag=as50289726 To: ;tag=6481f832 Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 103 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 17 23:00:55] DEBUG[4986] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #31 [Jan 17 23:00:55] DEBUG[4986] chan_sip.c: Trying to put 'BYE sip:*11' onto UDP socket destined for 192.168.15.5:58988 [Jan 17 23:00:55] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *11 [Jan 17 23:00:55] DEBUG[4941] chan_sip.c: Checking device state for peer *11 [Jan 17 23:00:55] DEBUG[4941] devicestate.c: Changing state for SIP/*11 - state 1 (Not in use) [Jan 17 23:00:55] DEBUG[4941] devicestate.c: device 'SIP/*11' state '1' [Jan 17 23:00:55] DEBUG[4986] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 17 23:00:55] DEBUG[4986] pbx.c: Spawn extension (local,*11,1) exited non-zero on 'SIP/*10-00000000' [Jan 17 23:00:55] VERBOSE[4986] pbx.c: == Spawn extension (local, *11, 1) exited non-zero on 'SIP/*10-00000000' [Jan 17 23:00:55] DEBUG[4986] channel.c: Soft-Hanging up channel 'SIP/*10-00000000' [Jan 17 23:00:55] DEBUG[4986] channel.c: Hanging up channel 'SIP/*10-00000000' [Jan 17 23:00:55] DEBUG[4986] chan_sip.c: Hangup call SIP/*10-00000000, SIP callid ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:00:55] DEBUG[4986] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x87808c0' [Jan 17 23:00:55] DEBUG[4986] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8784b50' [Jan 17 23:00:55] DEBUG[4941] devicestate.c: No provider found, checking channel drivers for SIP - *10 [Jan 17 23:00:55] DEBUG[4941] chan_sip.c: Checking device state for peer *10 [Jan 17 23:00:55] DEBUG[4941] devicestate.c: Changing state for SIP/*10 - state 1 (Not in use) [Jan 17 23:00:55] DEBUG[4941] devicestate.c: device 'SIP/*10' state '1' [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.615132','HANGUP','','','*11','','','','','local','SIP/*11-00000001','AppDial','(Outgoing Line)','','','1295301614.1','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.615737','CHAN_END','','','*11','','','','','local','SIP/*11-00000001','AppDial','(Outgoing Line)','','','1295301614.1','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] DEBUG[4978] app_queue.c: Device 'SIP/*11' changed to state '1' (Not in use) [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.616019','HANGUP','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','','','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.616110','CHAN_END','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','','','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] VERBOSE[4936] cel_pgsql.c: > [INSERT INTO cel ("id","eventtime","eventtype","userdeftype","cid_name","cid_num","cid_ani","cid_rdnis","cid_dnid","exten","context","channame","appname","appdata","accountcode","peeraccount","uniqueid","linkedid","amaflags","userfield","peer") VALUES (DEFAULT,'2011-01-17 23:00:55.616701','LINKEDID_END','','XLite','*10','*10','','*11','*11','local','SIP/*10-00000000','','','','','1295301614.0','1295301614.0',3,'','')] [Jan 17 23:00:55] DEBUG[4936] cel_pgsql.c: inserting a CEL record. [Jan 17 23:00:55] DEBUG[4978] app_queue.c: Device 'SIP/*10' changed to state '1' (Not in use) [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: <--- SIP read from UDP:192.168.15.5:58988 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK6a771751 Contact: To: ;tag=6481f832 From: "XLite";tag=as50289726 Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 CSeq: 103 BYE User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 Content-Length: 0 <-------------> [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.15.21:5060;branch=z9hG4bK6a771751 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 2 [ 64]: Contact: [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 3 [ 72]: To: ;tag=6481f832 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 4 [ 51]: From: "XLite";tag=as50289726 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 5 [ 60]: Call-ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 6 [ 13]: CSeq: 103 BYE [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 7 [ 50]: User-Agent: X-Lite Beta release 4.0 v3 stamp 55153 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: --- (9 headers 0 lines) --- [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: = Looking for Call ID: 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 (Checking To) --From tag as50289726 --To-tag 6481f832 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #31 [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Stopping retransmission on '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' of Request 103: Match Found [Jan 17 23:00:55] DEBUG[4953] chan_sip.c: Destroying SIP dialog 1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060 [Jan 17 23:00:55] VERBOSE[4953] chan_sip.c: Really destroying SIP dialog '1404f1f00ac30119338bc74f5def57c1@192.168.15.21:5060' Method: INVITE [Jan 17 23:00:55] DEBUG[4953] rtp_engine.c: Destroyed RTP instance '0x877c8b8' [Jan 17 23:00:55] DEBUG[4953] rtp_engine.c: Destroyed RTP instance '0x87765f8' [Jan 17 23:00:59] VERBOSE[4944] asterisk.c: -- Remote UNIX connection disconnected [Jan 17 23:01:02] DEBUG[4953] chan_sip.c: Auto destroying SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' [Jan 17 23:01:02] DEBUG[4953] chan_sip.c: Destroying SIP dialog ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM. [Jan 17 23:01:02] VERBOSE[4953] chan_sip.c: Really destroying SIP dialog 'ZmIyNmE2MzIxY2FjNDcwZjExZjYzYTdkNzdhOTI0MzM.' Method: BYE [Jan 17 23:01:02] DEBUG[4953] rtp_engine.c: Destroyed RTP instance '0x87808c0' [Jan 17 23:01:02] DEBUG[4953] rtp_engine.c: Destroyed RTP instance '0x8784b50'