[Jan 10 17:55:30] Asterisk Ready. ]1;Asterisk]2;Asterisk Console on 'srv525' (pid 14565)*CLI> [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.3b5e Our tag: as5c06758c [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 7e54ba3c5bceecf534292c402b522ee9@84.234.24.35 Their Tag Our tag: as5c790220 [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.98d1 Our tag: as0cc366ed [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: 76df03fe3fff1bac4fe7b4b26b61e8cd@84.234.24.35 Their Tag Our tag: as6ced0a44 [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '76df03fe3fff1bac4fe7b4b26b61e8cd@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:55:30] NOTICE[14576]: chan_sip.c:12001 handle_response_peerpoke: Peer '1100' is now Reachable. (155ms / 2000ms) [Jan 10 17:55:30] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1100 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1100 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1100 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1100 - state 1 (Not in use) [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1100 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1100 [Jan 10 17:55:30] Really destroying SIP dialog '76df03fe3fff1bac4fe7b4b26b61e8cd@84.234.24.35' Method: OPTIONS [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: 0baa87c14ca816464361842767809b1f@84.234.24.35 Their Tag Our tag: as660037f3 [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0baa87c14ca816464361842767809b1f@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:55:30] NOTICE[14576]: chan_sip.c:12001 handle_response_peerpoke: Peer '1001' is now Reachable. (43ms / 2000ms) [Jan 10 17:55:30] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 1 (Not in use) [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:55:30] Really destroying SIP dialog '0baa87c14ca816464361842767809b1f@84.234.24.35' Method: OPTIONS [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: 2ba1e4067f4daed11c7b29e97c389c89@84.234.24.35 Their Tag Our tag: as3709d51f [Jan 10 17:55:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '2ba1e4067f4daed11c7b29e97c389c89@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:55:30] NOTICE[14576]: chan_sip.c:12001 handle_response_peerpoke: Peer '1000' is now Reachable. (57ms / 2000ms) [Jan 10 17:55:30] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 1 (Not in use) [Jan 10 17:55:30] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:55:30] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:55:30] Really destroying SIP dialog '2ba1e4067f4daed11c7b29e97c389c89@84.234.24.35' Method: OPTIONS [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.3b5e Our tag: as5c06758c [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 7e54ba3c5bceecf534292c402b522ee9@84.234.24.35 Their Tag Our tag: as5c790220 [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.98d1 Our tag: as0cc366ed [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 3548c80d47e440e75535ef8152f2e7d4@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a0e4 Our tag: as1d2d7b18 [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a931 Our tag: as26ad85fb [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35 Their Tag Our tag: as012d1192 [Jan 10 17:55:31] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:55:32] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:34] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:34] NOTICE[14576]: chan_sip.c:14934 sip_poke_noanswer: Peer '1003' is now UNREACHABLE! Last qualify: 0 [Jan 10 17:55:34] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 7e54ba3c5bceecf534292c402b522ee9@84.234.24.35 [Jan 10 17:55:34] Really destroying SIP dialog '7e54ba3c5bceecf534292c402b522ee9@84.234.24.35' Method: OPTIONS [Jan 10 17:55:34] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:34] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:34] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:34] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:55:34] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:34] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:38] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.3b5e Our tag: as5c06758c [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.98d1 Our tag: as0cc366ed [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 3548c80d47e440e75535ef8152f2e7d4@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a0e4 Our tag: as1d2d7b18 [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a931 Our tag: as26ad85fb [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35 Their Tag Our tag: as012d1192 [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. - REGISTER (No RTP) [Jan 10 17:55:42] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 10 17:55:44] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) *CLI> [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #35)) [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 70f2c4425b9fce2a25ea93b118e47030@84.234.24.35 Their Tag Our tag: as64e524b0 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14472 handle_request: Ignoring SIP message because of retransmit (REGISTER Seqno 1, ours 1) set[Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 70f2c4425b9fce2a25ea93b118e47030@84.234.24.35 Their Tag Our tag: as64e524b0 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 70f2c4425b9fce2a25ea93b118e47030@84.234.24.35 [Jan 10 17:55:46] Really destroying SIP dialog '70f2c4425b9fce2a25ea93b118e47030@84.234.24.35' Method: OPTIONS [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:55:46] -- Registered SIP '1003' at 83.105.95.134 port 44780 expires 3600 [Jan 10 17:55:46] -- Saved useragent "X-Lite release 1006e stamp 34025" for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: 1e52c4560e39002707f7bb8d488eabd9@84.234.24.35 Their Tag Our tag: as70fc884d [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1e52c4560e39002707f7bb8d488eabd9@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:55:46] NOTICE[14576]: chan_sip.c:12001 handle_response_peerpoke: Peer '1003' is now Reachable. (41ms / 2000ms) [Jan 10 17:55:46] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:46] Really destroying SIP dialog '1e52c4560e39002707f7bb8d488eabd9@84.234.24.35' Method: OPTIONS [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 10 17:55:46] -- Unregistered SIP '1003' [Jan 10 17:55:46] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 5 (Unavailable) [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.3b5e Our tag: as5c06758c [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.98d1 Our tag: as0cc366ed [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 3548c80d47e440e75535ef8152f2e7d4@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a0e4 Our tag: as1d2d7b18 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a931 Our tag: as26ad85fb [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35 Their Tag Our tag: as012d1192 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:55:46] -- Registered SIP '1003' at 83.105.95.134 port 44780 expires 3600 [Jan 10 17:55:46] -- Saved useragent "X-Lite release 1006e stamp 34025" for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: 6fde3102517a5e47289b35655df424f7@84.234.24.35 Their Tag Our tag: as22f58a96 [Jan 10 17:55:46] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6fde3102517a5e47289b35655df424f7@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:55:46] NOTICE[14576]: chan_sip.c:12001 handle_response_peerpoke: Peer '1003' is now Reachable. (48ms / 2000ms) [Jan 10 17:55:46] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:55:46] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:55:46] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:55:46] Really destroying SIP dialog '6fde3102517a5e47289b35655df424f7@84.234.24.35' Method: OPTIONS [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. Their Tag 5d0f554c Our tag: as55e2d3c7 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.3b5e Our tag: as5c06758c [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.98d1 Our tag: as0cc366ed [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 3548c80d47e440e75535ef8152f2e7d4@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a0e4 Our tag: as1d2d7b18 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35 Their Tag 13724058500c4c29465e0085181d997d.a931 Our tag: as26ad85fb [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = No match Their Call ID: 129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35 Their Tag Our tag: as012d1192 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY. - SUBSCRIBE (No RTP) [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY. Their Tag b149ec17 Our tag: as141cc0d2 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:14260 handle_request_subscribe: Adding subscription for mailbox notification - peer 1003 Mailbox 1000@vm_all [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:1862 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 96 ms (t1 48 ms (Retrans id #95)) [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY. Their Tag b149ec17 Our tag: as141cc0d2 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on 'ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY.' of Request 102: Match Not Found [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:4315 find_call: = Found Their Call ID: ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY. Their Tag b149ec17 Our tag: as141cc0d2 [Jan 10 17:55:47] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on 'ZGQxODAyY2M0OTU3MDkwZDZmYjRiM2ZmODk2MTE1YjY.' of Request 102: Match Found No such command 'set' (type 'help' for help) *CLI> *CLI> set debug 4 Core debug was 5 and is now 4 The 'set debug' command is deprecated and will be removed in a future release. Please use 'core set debug' instead. *CLI> set verbose 4 Verbosity was 5 and is now 4 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. *CLI> set [Jan 10 17:56:01] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35' [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35 [Jan 10 17:56:02] Really destroying SIP dialog '129b78da0d51ae9a7dbcf8522feb10c7@84.234.24.35' Method: NOTIFY [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35' [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35 [Jan 10 17:56:02] Really destroying SIP dialog '0a2be9416c6e13f526ff25bb09d87f96@84.234.24.35' Method: REGISTER [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '3548c80d47e440e75535ef8152f2e7d4@84.234.24.35' [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 3548c80d47e440e75535ef8152f2e7d4@84.234.24.35 [Jan 10 17:56:02] Really destroying SIP dialog '3548c80d47e440e75535ef8152f2e7d4@84.234.24.35' Method: REGISTER [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35' [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35 [Jan 10 17:56:02] Really destroying SIP dialog '6d7e63891d262e2f6aad09e84a4ac586@84.234.24.35' Method: REGISTER [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '596c8eff4024569a7f6cb25a0abb5929@84.234.24.35' [Jan 10 17:56:02] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 596c8eff4024569a7f6cb25a0abb5929@84.234.24.35 [Jan 10 17:56:02] Really destroying SIP dialog '596c8eff4024569a7f6cb25a0abb5929@84.234.24.35' Method: REGISTER  *CLI> sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> sip debug[Jan 10 17:56:12] <--- SIP read from 90.195.58.25:50518 ---> <-------------> [Jan 10 17:56:12] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: (0) [Jan 10 17:56:12] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: (0) [Jan 10 17:56:12] --- (0 headers 1 lines) --- on Usage: sip set debug Enables dumping of SIP packets for debugging purposes sip set debug ip Enables dumping of SIP packets to and from host. sip set debug peer Enables dumping of SIP packets to and from host. Require peer to be registered. *CLI> [Jan 10 17:56:16] <--- SIP read from 83.105.95.134:44780 ---> <-------------> [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: (0) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: (0) [Jan 10 17:56:16] --- (0 headers 1 lines) --- [Jan 10 17:56:16] <--- SIP read from 89.213.46.56:64435 ---> NOTIFY sip:84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b645e7de From: "James Brindle" ;tag=16273a752963b43co3 To: Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1821 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: NOTIFY sip:84.234.24.35 SIP/2.0 (31) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b645e7de (59) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James Brindle" ;tag=16273a752963b43co3 (68) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (22) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 448cad21-a7370180@192.168.1.211 (40) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1821 NOTIFY (17) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Event: keep-alive (17) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Linksys/SPA941-4.1.15 (33) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:56:16] --- (10 headers 0 lines) --- [Jan 10 17:56:16] <--- Transmitting (no NAT) to 89.213.46.56:64435 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b645e7de;received=89.213.46.56 From: "James Brindle" ;tag=16273a752963b43co3 To: ;tag=as71fc57d0 Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1821 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 10 17:56:16] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:56:18] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog 'ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg.' [Jan 10 17:56:18] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg. [Jan 10 17:56:18] Really destroying SIP dialog 'ZWUyMTY3ODFkNzMwNzJhZDhjZmU0ZTBmYmJiOTJhODg.' Method: REGISTER [Jan 10 17:56:29] <--- SIP read from 217.14.132.178:5060 ---> INVITE sip:8001@84.234.24.35 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK2dc5.71a76c97.0 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK1d8649a1;rport=5060 From: "07870920797" ;tag=as03fe6293 To: Contact: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 102 INVITE User-Agent: Telappliant VoIP Gateway Max-Forwards: 16 Date: Wed, 10 Jan 2007 17:56:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 336 v=0 o=root 4723 4723 IN IP4 217.14.138.48 s=session c=IN IP4 217.14.138.48 t=0 0 m=audio 24996 RTP/AVP 8 0 97 3 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:8001@84.234.24.35 SIP/2.0 (36) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Record-Route: (56) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK2dc5.71a76c97.0 (61) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK1d8649a1;rport=5060 (69) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "07870920797" ;tag=as03fe6293 (67) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: To: (36) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Contact: (40) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 (56) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Telappliant VoIP Gateway (36) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 16 (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Date: Wed, 10 Jan 2007 17:56:36 GMT (35) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: Content-Type: application/sdp (29) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 14: Content-Length: 336 (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 15: (0) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=root 4723 4723 IN IP4 217.14.138.48 (37) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=session (9) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 217.14.138.48 (22) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 24996 RTP/AVP 8 0 97 3 18 101 (37) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:18 annexb=no (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 10 17:56:29] --- (15 headers 15 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 - INVITE (With RTP) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:29] Sending to 217.14.132.178 : 5060 (no NAT) [Jan 10 17:56:29] Using INVITE request as basis request - 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 [Jan 10 17:56:29] Found peer 'HALLNET_OUT' [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Jan 10 17:56:29] Found RTP audio format 8 [Jan 10 17:56:29] Found RTP audio format 0 [Jan 10 17:56:29] Found RTP audio format 97 [Jan 10 17:56:29] Found RTP audio format 3 [Jan 10 17:56:29] Found RTP audio format 18 [Jan 10 17:56:29] Found RTP audio format 101 [Jan 10 17:56:29] Peer audio RTP is at port 217.14.138.48:24996 [Jan 10 17:56:29] Found description format PCMA for ID 8 [Jan 10 17:56:29] Found description format PCMU for ID 0 [Jan 10 17:56:29] Found description format iLBC for ID 97 [Jan 10 17:56:29] Found description format GSM for ID 3 [Jan 10 17:56:29] Found description format G729 for ID 18 [Jan 10 17:56:29] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:29] Found description format telephone-event for ID 101 [Jan 10 17:56:29] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel [Jan 10 17:56:29] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:29] Peer audio RTP is at port 217.14.138.48:24996 [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:13237 handle_request_invite: Checking SIP call limits for device 7169832 [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Jan 10 17:56:29] Looking for 8001 in incoming (domain 84.234.24.35) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:7862 build_route: build_route: Record-Route hop: [Jan 10 17:56:29] list_route: hop: [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:13310 handle_request_invite: SIP/7169832-081bec40: New call is still down.... Trying... [Jan 10 17:56:29] <--- Transmitting (no NAT) to 217.14.132.178:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK2dc5.71a76c97.0;received=217.14.132.178 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK1d8649a1;rport=5060 From: "07870920797" ;tag=as03fe6293 To: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:29] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/7169832-081bec40 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 7169832 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 7169832 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/7169832 - state 4 (Invalid) [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1767 pbx_extension_helper: Launching 'LookupCIDName' [Jan 10 17:56:29] -- Executing [8001@incoming:1] LookupCIDName("SIP/7169832-081bec40", "") in new stack [Jan 10 17:56:29] WARNING[14594]: app_lookupcidname.c:70 lookupcidname_exec: LookupCIDName is deprecated. Please use ${DB(cidname/${CALLERID(num)})} instead. [Jan 10 17:56:29] DEBUG[14594]: db.c:197 ast_db_get: Unable to find key '07870920797' in family 'cidname' [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '07870920797' [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1767 pbx_extension_helper: Launching 'Set' [Jan 10 17:56:29] -- Executing [8001@incoming:2] Set("SIP/7169832-081bec40", "CALLERID(num)="907870920797"") in new stack [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1621 pbx_substitute_variables_helper_full: Function result is '07870920797' [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1767 pbx_extension_helper: Launching 'Set' [Jan 10 17:56:29] -- Executing [8001@incoming:3] Set("SIP/7169832-081bec40", "CALLERID(name)="JBC: 07870920797"") in new stack [Jan 10 17:56:29] DEBUG[14594]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' [Jan 10 17:56:29] -- Executing [8001@incoming:4] Dial("SIP/7169832-081bec40", "SIP/1000&SIP/1001&SIP/1002&SIP/1003|30|t") in new stack [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-4. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-3. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-2. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-1. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2812 sip_call: Outgoing Call for 1000 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3048 update_call_counter: Call to peer '1000' is 1 out of 200 [Jan 10 17:56:29] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:29] Audio is at 84.234.24.35 port 16384 [Jan 10 17:56:29] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:29] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:29] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:29] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1000@192.168.1.211:5063 SIP/2.0 (42) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6;rport (63) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as617d0f40 (71) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 3: To: (33) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 4: Contact: (40) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:29 GMT (35) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 287 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: o=root 14594 14594 IN IP4 84.234.24.35 (38) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: s=session (9) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: m=audio 16384 RTP/AVP 0 8 3 101 (31) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:29] Reliably Transmitting (NAT) to 89.213.46.56:64435: INVITE sip:1000@192.168.1.211:5063 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6;rport From: "JBC: 07870920797" ;tag=as617d0f40 To: Contact: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 14594 14594 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 16384 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #96 [Jan 10 17:56:29] -- Called 1000 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-4. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-3. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-2. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-1. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2812 sip_call: Outgoing Call for 1001 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3048 update_call_counter: Call to peer '1001' is 1 out of 200 [Jan 10 17:56:29] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:29] Audio is at 84.234.24.35 port 18604 [Jan 10 17:56:29] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:29] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:29] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:29] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1001@10.0.0.213:3876 SIP/2.0 (39) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport (63) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 3: To: (30) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 4: Contact: (40) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:29 GMT (35) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 287 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: o=root 14594 14594 IN IP4 84.234.24.35 (38) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: s=session (9) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: m=audio 18604 RTP/AVP 0 8 3 101 (31) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:29] Reliably Transmitting (NAT) to 83.105.95.134:43460: INVITE sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport From: "JBC: 07870920797" ;tag=as66e05052 To: Contact: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 14594 14594 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 18604 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #98 [Jan 10 17:56:29] -- Called 1001 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:15136 sip_request_call: Cant create SIP call - target device not registred [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 05135d450d0b18d25599c4ab1c7d2ac1@84.234.24.35 [Jan 10 17:56:29] Really destroying SIP dialog '05135d450d0b18d25599c4ab1c7d2ac1@84.234.24.35' Method: INVITE [Jan 10 17:56:29] WARNING[14594]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-4. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-3. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-2. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-incoming-8001-1. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 10 17:56:29] DEBUG[14594]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2812 sip_call: Outgoing Call for 1003 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:3048 update_call_counter: Call to peer '1003' is 1 out of 200 [Jan 10 17:56:29] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:29] Audio is at 84.234.24.35 port 5406 [Jan 10 17:56:29] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:29] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:29] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:29] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 (70) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport (63) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as160afd16 (71) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 3: To: (61) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 4: Contact: (40) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:29 GMT (35) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 286 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: o=root 14594 14594 IN IP4 84.234.24.35 (38) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: s=session (9) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: m=audio 5406 RTP/AVP 0 8 3 101 (30) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:29] Reliably Transmitting (NAT) to 83.105.95.134:44780: INVITE sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport From: "JBC: 07870920797" ;tag=as160afd16 To: Contact: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 286 v=0 o=root 14594 14594 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 5406 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 10 17:56:29] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #100 [Jan 10 17:56:29] -- Called 1003 [Jan 10 17:56:29] <--- SIP read from 89.213.46.56:64435 ---> SIP/2.0 100 Trying To: From: "JBC: 07870920797" ;tag=as617d0f40 Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 INVITE Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: To: (33) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as617d0f40 (71) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 (57) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Server: Linksys/SPA941-4.1.15 (29) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:29] --- (8 headers 0 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #96 - INVITE (got response) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' Request 102: Found [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 100 to standard invite [Jan 10 17:56:29] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport From: "JBC: 07870920797" ;tag=as66e05052 To: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 100 Trying (18) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport (63) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (30) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:29] --- (8 headers 0 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #98 - INVITE (got response) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' Request 102: Found [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 100 to standard invite [Jan 10 17:56:29] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport (63) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=f37978bef01506b5 (51) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (35) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:56:29] --- (10 headers 0 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' Request 102: Found [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Jan 10 17:56:29] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081d0fb0 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:29] -- SIP/1001-081d0fb0 is ringing [Jan 10 17:56:29] <--- Transmitting (no NAT) to 217.14.132.178:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK2dc5.71a76c97.0;received=217.14.132.178 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK1d8649a1;rport=5060 From: "07870920797" ;tag=as03fe6293 To: ;tag=as1e337580 Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:29] <--- SIP read from 89.213.46.56:64435 ---> SIP/2.0 180 Ringing To: ;tag=cf1827625b4d4daei3 From: "JBC: 07870920797" ;tag=as617d0f40 Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 INVITE Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=cf1827625b4d4daei3 (56) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as617d0f40 (71) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 (57) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Server: Linksys/SPA941-4.1.15 (29) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:29] --- (8 headers 0 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' Request 102: Found [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Jan 10 17:56:29] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000-081bd2a8 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:29] -- SIP/1000-081bd2a8 is ringing [Jan 10 17:56:29] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 Contact: To: ;tag=1f44393b From: "JBC: 07870920797";tag=as160afd16 Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 (68) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=1f44393b (74) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "JBC: 07870920797";tag=as160afd16 (70) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 (54) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: (0) [Jan 10 17:56:29] --- (9 headers 0 lines) --- [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #100 - INVITE (got response) [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' Request 102: Found [Jan 10 17:56:29] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Jan 10 17:56:29] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081d4f18 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 10 17:56:29] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:29] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:29] -- SIP/1003-081d4f18 is ringing [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1201@10.27.10.99:57072 SIP/2.0 (42) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK38e0e1a9;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as730a38be (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (32) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 43cf78100e1e6de91bcee7e910fffd08@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:30 GMT (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:30] Reliably Transmitting (NAT) to 82.33.171.11:57072: OPTIONS sip:1201@10.27.10.99:57072 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK38e0e1a9;rport From: "asterisk" ;tag=as730a38be To: Contact: Call-ID: 43cf78100e1e6de91bcee7e910fffd08@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #102 [Jan 10 17:56:30] <--- SIP read from 82.33.171.11:57072 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK38e0e1a9;rport From: "asterisk" ;tag=as730a38be To: ;tag=5dacc50e825e5d04 Call-ID: 43cf78100e1e6de91bcee7e910fffd08@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK38e0e1a9;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as730a38be (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=5dacc50e825e5d04 (53) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 43cf78100e1e6de91bcee7e910fffd08@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (37) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Supported: replaces, timer (26) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:30] --- (11 headers 0 lines) --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #102 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '43cf78100e1e6de91bcee7e910fffd08@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:30] Really destroying SIP dialog '43cf78100e1e6de91bcee7e910fffd08@84.234.24.35' Method: OPTIONS [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1200@10.0.1.102:46850 SIP/2.0 (41) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK02f1f1fa;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as337e9e71 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (31) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 47548d0243e78f425cf5bb6b723c841d@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:30 GMT (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:30] Reliably Transmitting (NAT) to 80.176.225.214:46850: OPTIONS sip:1200@10.0.1.102:46850 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK02f1f1fa;rport From: "asterisk" ;tag=as337e9e71 To: Contact: Call-ID: 47548d0243e78f425cf5bb6b723c841d@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #105 [Jan 10 17:56:30] <--- SIP read from 80.176.225.214:46850 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK02f1f1fa;rport From: "asterisk" ;tag=as337e9e71 To: ;tag=e147e4bf61544f60 Call-ID: 47548d0243e78f425cf5bb6b723c841d@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK02f1f1fa;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as337e9e71 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=e147e4bf61544f60 (52) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 47548d0243e78f425cf5bb6b723c841d@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Supported: replaces, timer (26) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:30] --- (11 headers 0 lines) --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #105 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '47548d0243e78f425cf5bb6b723c841d@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:30] Really destroying SIP dialog '47548d0243e78f425cf5bb6b723c841d@84.234.24.35' Method: OPTIONS [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1100@90.195.58.25:50518;rinstance=a0480cdfe9737f15 SIP/2.0 (70) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK074cf985;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as4491ada5 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (60) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:30 GMT (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:30] Reliably Transmitting (NAT) to 90.195.58.25:50518: OPTIONS sip:1100@90.195.58.25:50518;rinstance=a0480cdfe9737f15 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK074cf985;rport From: "asterisk" ;tag=as4491ada5 To: Contact: Call-ID: 0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #108 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1001@10.0.0.213:3876 SIP/2.0 (40) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK3e3f9730;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as01b0ef70 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (30) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:30 GMT (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:30] Reliably Transmitting (NAT) to 83.105.95.134:43460: OPTIONS sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK3e3f9730;rport From: "asterisk" ;tag=as01b0ef70 To: Contact: Call-ID: 1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #110 [Jan 10 17:56:30] <--- SIP read from 90.195.58.25:50518 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK074cf985;rport=5060 Contact: To: ;tag=16005433 From: "asterisk";tag=as4491ada5 Call-ID: 0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35 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 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK074cf985;rport=5060 (68) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (32) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=16005433 (73) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "asterisk";tag=as4491ada5 (58) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Accept: application/sdp (23) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Accept-Language: en (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:30] --- (12 headers 0 lines) --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #108 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:30] Really destroying SIP dialog '0b356b140bc7b97a2c88ca0f33980d2d@84.234.24.35' Method: OPTIONS [Jan 10 17:56:30] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK3e3f9730;rport From: "asterisk" ;tag=as01b0ef70 To: ;tag=f37978bef01506b5 Call-ID: 1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK3e3f9730;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as01b0ef70 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=f37978bef01506b5 (51) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Supported: replaces, timer (26) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:30] --- (11 headers 0 lines) --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #110 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:30] Really destroying SIP dialog '1b2ca80b7eb0532e08d991940bf57c1a@84.234.24.35' Method: OPTIONS [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1000@192.168.1.211:5063 SIP/2.0 (43) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK688230ae;rport (63) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as606acf45 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (33) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 34c752664cfffdb1082e33923fbe8308@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:30 GMT (35) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:30] Reliably Transmitting (NAT) to 89.213.46.56:64435: OPTIONS sip:1000@192.168.1.211:5063 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK688230ae;rport From: "asterisk" ;tag=as606acf45 To: Contact: Call-ID: 34c752664cfffdb1082e33923fbe8308@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #114 [Jan 10 17:56:30] <--- SIP read from 89.213.46.56:64435 ---> SIP/2.0 200 OK To: ;tag=a6beeda933d27a98i3 From: "asterisk" ;tag=as606acf45 Call-ID: 34c752664cfffdb1082e33923fbe8308@84.234.24.35 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK688230ae Server: Linksys/SPA941-4.1.15 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER <-------------> [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=a6beeda933d27a98i3 (56) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as606acf45 (59) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 34c752664cfffdb1082e33923fbe8308@84.234.24.35 (54) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK688230ae (57) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Server: Linksys/SPA941-4.1.15 (29) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: (0) [Jan 10 17:56:30] --- (9 headers 0 lines) --- [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #114 [Jan 10 17:56:30] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '34c752664cfffdb1082e33923fbe8308@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:30] Really destroying SIP dialog '34c752664cfffdb1082e33923fbe8308@84.234.24.35' Method: OPTIONS [Jan 10 17:56:31] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 102 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer Content-Length: 207 v=0 o=1001 8000 8000 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 5954 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK2439fca6;rport (63) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=f37978bef01506b5 (51) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (35) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Type: application/sdp (29) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Supported: replaces, timer (26) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 207 (19) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8000 8000 IN IP4 10.0.0.213 (34) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5954 RTP/AVP 0 101 (26) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:31] --- (12 headers 11 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Jan 10 17:56:31] Found RTP audio format 0 [Jan 10 17:56:31] Found RTP audio format 101 [Jan 10 17:56:31] Peer audio RTP is at port 10.0.0.213:5954 [Jan 10 17:56:31] Found description format PCMU for ID 0 [Jan 10 17:56:31] Found description format telephone-event for ID 101 [Jan 10 17:56:31] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1001-081d0fb0 [Jan 10 17:56:31] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Jan 10 17:56:31] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:31] Peer audio RTP is at port 10.0.0.213:5954 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0x4 (ulaw) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:31] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:7891 build_route: build_route: Contact hop: [Jan 10 17:56:31] list_route: hop: [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:5576 reqprep: Strict routing enforced for session 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:31] set_destination: Parsing for address/port to send to [Jan 10 17:56:31] set_destination: set destination to 10.0.0.213, port 3876 [Jan 10 17:56:31] Transmitting (NAT) to 83.105.95.134:43460: ACK sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK47e6d878;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Contact: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081d0fb0 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:31] -- SIP/1001-081d0fb0 answered SIP/7169832-081bec40 [Jan 10 17:56:31] DEBUG[14594]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/1003-081d4f18' [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3278 sip_hangup: Hangup call SIP/1003-081d4f18, SIP callid 3810ed2b60ac22913ea04788789ae20b@84.234.24.35) [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3286 sip_hangup: update_call_counter(1003) - decrement call limit counter on hangup [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3022 update_call_counter: Call to peer '1003' removed from call limit 200 [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3299 sip_hangup: Hanging up channel in state Ringing (not UP) [Jan 10 17:56:31] Scheduling destruction of SIP dialog '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' in 6400 ms (Method: INVITE) [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:31] Reliably Transmitting (NAT) to 83.105.95.134:44780: CANCEL sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport From: "JBC: 07870920797" ;tag=as160afd16 To: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #118 [Jan 10 17:56:31] Scheduling destruction of SIP dialog '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' in 6400 ms (Method: INVITE) [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081d4f18 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14594]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/1000-081bd2a8' [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3278 sip_hangup: Hangup call SIP/1000-081bd2a8, SIP callid 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35) [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3286 sip_hangup: update_call_counter(1000) - decrement call limit counter on hangup [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3022 update_call_counter: Call to peer '1000' removed from call limit 200 [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3299 sip_hangup: Hanging up channel in state Ringing (not UP) [Jan 10 17:56:31] Scheduling destruction of SIP dialog '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' in 6400 ms (Method: INVITE) [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:31] Reliably Transmitting (NAT) to 89.213.46.56:64435: CANCEL sip:1000@192.168.1.211:5063 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6;rport From: "JBC: 07870920797" ;tag=as617d0f40 To: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #121 [Jan 10 17:56:31] Scheduling destruction of SIP dialog '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' in 6400 ms (Method: INVITE) [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000-081bd2a8 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/7169832-081bec40 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 7169832 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 7169832 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/7169832 - state 4 (Invalid) [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:3428 sip_answer: SIP answering channel: SIP/7169832-081bec40 [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 10 17:56:31] Audio is at 84.234.24.35 port 5792 [Jan 10 17:56:31] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:31] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:31] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:31] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:31] <--- Reliably Transmitting (no NAT) to 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK2dc5.71a76c97.0;received=217.14.132.178 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK1d8649a1;rport=5060 Record-Route: From: "07870920797" ;tag=as03fe6293 To: ;tag=as1e337580 Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 286 v=0 o=root 14594 14594 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 5792 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 10 17:56:31] DEBUG[14594]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #123 [Jan 10 17:56:31] <--- SIP read from 217.14.132.178:5060 ---> ACK sip:8001@84.234.24.35 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.14.132.178;branch=0 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK4a34a732;rport=5060 From: "07870920797" ;tag=as03fe6293 To: ;tag=as1e337580 Contact: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 102 ACK User-Agent: Telappliant VoIP Gateway Max-Forwards: 16 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:8001@84.234.24.35 SIP/2.0 (33) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Record-Route: (56) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 217.14.132.178;branch=0 (40) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK4a34a732;rport=5060 (69) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "07870920797" ;tag=as03fe6293 (67) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as1e337580 (51) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Contact: (40) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 (56) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: CSeq: 102 ACK (13) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Telappliant VoIP Gateway (36) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 16 (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:31] --- (12 headers 0 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #123 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '7e69b30d05593e6a1482ff6a404075e5@217.14.132.185' of Response 102: Match Not Found [Jan 10 17:56:31] DEBUG[14594]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:45538 [Jan 10 17:56:31] DEBUG[14594]: rtp.c:2647 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 10 17:56:31] DEBUG[14594]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:31] <--- SIP read from 89.213.46.56:64435 ---> SIP/2.0 487 Request Terminated To: ;tag=cf1827625b4d4daei3 From: "JBC: 07870920797" ;tag=as617d0f40 Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 INVITE Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=cf1827625b4d4daei3 (56) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as617d0f40 (71) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 (54) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: CSeq: 102 INVITE (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 (57) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Server: Linksys/SPA941-4.1.15 (29) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:31] --- (8 headers 0 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' of Request 102: Match Found [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 487 to standard invite [Jan 10 17:56:31] Transmitting (NAT) to 89.213.46.56:64435: ACK sip:1000@192.168.1.211:5063 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6;rport From: "JBC: 07870920797" ;tag=as617d0f40 To: ;tag=cf1827625b4d4daei3 Contact: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:3022 update_call_counter: Call to peer '1000' removed from call limit 200 [Jan 10 17:56:31] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1000 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1000 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1000 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1000 [Jan 10 17:56:31] <--- SIP read from 89.213.46.56:64435 ---> SIP/2.0 200 OK To: ;tag=cf1827625b4d4daei3 From: "JBC: 07870920797" ;tag=as617d0f40 Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 CSeq: 102 CANCEL Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: To: ;tag=cf1827625b4d4daei3 (56) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as617d0f40 (71) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Call-ID: 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 (54) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: CSeq: 102 CANCEL (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK46ce08d6 (57) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Server: Linksys/SPA941-4.1.15 (29) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:31] --- (8 headers 0 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #121 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:31] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 Contact: To: ;tag=1f44393b From: "JBC: 07870920797";tag=as160afd16 Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 CANCEL User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 (68) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=1f44393b (74) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "JBC: 07870920797";tag=as160afd16 (70) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 (54) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 CANCEL (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: (0) [Jan 10 17:56:31] --- (9 headers 0 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #118 [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:31] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 To: ;tag=1f44393b From: "JBC: 07870920797";tag=as160afd16 Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 487 Request Terminated (30) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport=5060 (68) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: To: ;tag=1f44393b (74) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: From: "JBC: 07870920797";tag=as160afd16 (70) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 (54) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 INVITE (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:31] --- (8 headers 0 lines) --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' of Request 102: Match Found [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 487 to standard invite [Jan 10 17:56:31] Transmitting (NAT) to 83.105.95.134:44780: ACK sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK5bff02ed;rport From: "JBC: 07870920797" ;tag=as160afd16 To: ;tag=1f44393b Contact: Call-ID: 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:3022 update_call_counter: Call to peer '1003' removed from call limit 200 [Jan 10 17:56:31] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:56:31] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:31] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:31] DEBUG[14594]: rtp.c:2647 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 10 17:56:31] DEBUG[14594]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:31] <--- SIP read from 89.213.46.56:64435 ---> NOTIFY sip:84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-955887ae From: "James Brindle" ;tag=16273a752963b43co3 To: Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1822 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: NOTIFY sip:84.234.24.35 SIP/2.0 (31) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-955887ae (59) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James Brindle" ;tag=16273a752963b43co3 (68) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (22) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 448cad21-a7370180@192.168.1.211 (40) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1822 NOTIFY (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Event: keep-alive (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Linksys/SPA941-4.1.15 (33) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:56:31] --- (10 headers 0 lines) --- [Jan 10 17:56:31] <--- Transmitting (no NAT) to 89.213.46.56:64435 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-955887ae;received=89.213.46.56 From: "James Brindle" ;tag=16273a752963b43co3 To: ;tag=as412e11a4 Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1822 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 10 17:56:31] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:56:33] <--- SIP read from 83.105.95.134:43460 ---> INVITE sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK95cdc3bf09100ad3 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Contact: Supported: replaces, timer Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64435 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8000 8001 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 5954 RTP/AVP 0 3 4 8 18 2 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:907870920797@84.234.24.35 SIP/2.0 (44) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK95cdc3bf09100ad3 (63) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces, timer (26) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: CSeq: 64435 INVITE (18) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 331 (19) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: (0) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8000 8001 IN IP4 10.0.0.213 (34) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5954 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:33] --- (13 headers 16 lines) --- [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1662 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1670 parse_sip_options: Found SIP option: -replaces- [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1676 parse_sip_options: Matched SIP option: replaces [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1670 parse_sip_options: Found SIP option: -timer- [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1676 parse_sip_options: Matched SIP option: timer [Jan 10 17:56:33] Sending to 83.105.95.134 : 43460 (NAT) [Jan 10 17:56:33] Found RTP audio format 0 [Jan 10 17:56:33] Found RTP audio format 3 [Jan 10 17:56:33] Found RTP audio format 4 [Jan 10 17:56:33] Found RTP audio format 8 [Jan 10 17:56:33] Found RTP audio format 18 [Jan 10 17:56:33] Found RTP audio format 2 [Jan 10 17:56:33] Found RTP audio format 101 [Jan 10 17:56:33] Peer audio RTP is at port 10.0.0.213:5954 [Jan 10 17:56:33] Found description format PCMU for ID 0 [Jan 10 17:56:33] Found description format GSM for ID 3 [Jan 10 17:56:33] Found description format G723 for ID 4 [Jan 10 17:56:33] Found description format PCMA for ID 8 [Jan 10 17:56:33] Found description format G729 for ID 18 [Jan 10 17:56:33] Found description format G726-32 for ID 2 [Jan 10 17:56:33] Found description format telephone-event for ID 101 [Jan 10 17:56:33] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1001-081d0fb0 [Jan 10 17:56:33] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:33] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:33] Peer audio RTP is at port 10.0.0.213:5954 [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:13383 handle_request_invite: SIP/1001-081d0fb0: This call is UP.... [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:33] Audio is at 84.234.24.35 port 18604 [Jan 10 17:56:33] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:33] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:33] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:33] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:33] <--- Reliably Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK95cdc3bf09100ad3;received=83.105.95.134 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64435 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 14594 14595 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 18604 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #126 [Jan 10 17:56:33] -- Started music on hold, class 'default', on SIP/7169832-081bec40 [Jan 10 17:56:33] DEBUG[14594]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 10 17:56:33] DEBUG[14594]: channel.c:2225 __ast_read: Generator got voice, switching to phase locked mode [Jan 10 17:56:33] DEBUG[14594]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 10 17:56:33] DEBUG[14594]: channel.c:2731 set_format: Set channel SIP/7169832-081bec40 to write format slin [Jan 10 17:56:33] DEBUG[14594]: res_musiconhold.c:250 ast_moh_files_next: SIP/7169832-081bec40 Opened file 2 '/var/lib/asterisk/moh/fpm-world-mix' [Jan 10 17:56:33] <--- SIP read from 83.105.95.134:43460 ---> ACK sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK25311870d9d10cbe From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Contact: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64435 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:907870920797@84.234.24.35 SIP/2.0 (41) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK25311870d9d10cbe (63) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 64435 ACK (15) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (21) [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:33] --- (11 headers 0 lines) --- [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #126 [Jan 10 17:56:33] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' of Response 64435: Match Not Found [Jan 10 17:56:34] DEBUG[14594]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:45538 [Jan 10 17:56:34] DEBUG[14594]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:34] <--- SIP read from 83.105.95.134:43460 ---> INVITE sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK402f7ef27ee0c39d From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Contact: Supported: replaces, timer Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64436 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 328 v=0 o=1001 8000 8002 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 5954 RTP/AVP 0 3 4 8 18 2 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:907870920797@84.234.24.35 SIP/2.0 (44) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK402f7ef27ee0c39d (63) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces, timer (26) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: CSeq: 64436 INVITE (18) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 328 (19) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: (0) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8000 8002 IN IP4 10.0.0.213 (34) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 0.0.0.0 (16) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5954 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:34] --- (13 headers 16 lines) --- [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:34] Sending to 83.105.95.134 : 43460 (NAT) [Jan 10 17:56:34] Found RTP audio format 0 [Jan 10 17:56:34] Found RTP audio format 3 [Jan 10 17:56:34] Found RTP audio format 4 [Jan 10 17:56:34] Found RTP audio format 8 [Jan 10 17:56:34] Found RTP audio format 18 [Jan 10 17:56:34] Found RTP audio format 2 [Jan 10 17:56:34] Found RTP audio format 101 [Jan 10 17:56:34] Peer audio RTP is at port 0.0.0.0:5954 [Jan 10 17:56:34] Found description format PCMU for ID 0 [Jan 10 17:56:34] Found description format GSM for ID 3 [Jan 10 17:56:34] Found description format G723 for ID 4 [Jan 10 17:56:34] Found description format PCMA for ID 8 [Jan 10 17:56:34] Found description format G729 for ID 18 [Jan 10 17:56:34] Found description format G726-32 for ID 2 [Jan 10 17:56:34] Found description format telephone-event for ID 101 [Jan 10 17:56:34] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1001-081d0fb0 [Jan 10 17:56:34] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:34] Peer audio RTP is at port 0.0.0.0:5954 [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:13383 handle_request_invite: SIP/1001-081d0fb0: This call is UP.... [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:34] Audio is at 84.234.24.35 port 18604 [Jan 10 17:56:34] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:34] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:34] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:34] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:34] <--- Reliably Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK402f7ef27ee0c39d;received=83.105.95.134 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64436 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 14594 14596 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 18604 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jan 10 17:56:34] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #128 [Jan 10 17:56:34] -- Stopped music on hold on SIP/7169832-081bec40 [Jan 10 17:56:34] DEBUG[14594]: channel.c:2731 set_format: Set channel SIP/7169832-081bec40 to write format ulaw [Jan 10 17:56:34] -- Started music on hold, class 'default', on SIP/7169832-081bec40 [Jan 10 17:56:34] DEBUG[14594]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 10 17:56:34] DEBUG[14594]: channel.c:2225 __ast_read: Generator got voice, switching to phase locked mode [Jan 10 17:56:34] DEBUG[14594]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 10 17:56:34] DEBUG[14594]: channel.c:2731 set_format: Set channel SIP/7169832-081bec40 to write format slin [Jan 10 17:56:34] DEBUG[14594]: res_musiconhold.c:250 ast_moh_files_next: SIP/7169832-081bec40 Opened file 0 '/var/lib/asterisk/moh/fpm-calm-river' [Jan 10 17:56:35] <--- SIP read from 83.105.95.134:43460 ---> ACK sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK2e3770ab9c6df57d From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Contact: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64436 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:907870920797@84.234.24.35 SIP/2.0 (41) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK2e3770ab9c6df57d (63) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 64436 ACK (15) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (21) [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:35] --- (11 headers 0 lines) --- [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #128 [Jan 10 17:56:35] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' of Response 64436: Match Not Found [Jan 10 17:56:36] DEBUG[14594]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Jan 10 17:56:37] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' [Jan 10 17:56:37] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 3810ed2b60ac22913ea04788789ae20b@84.234.24.35 [Jan 10 17:56:37] Really destroying SIP dialog '3810ed2b60ac22913ea04788789ae20b@84.234.24.35' Method: INVITE [Jan 10 17:56:37] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' [Jan 10 17:56:37] DEBUG[14576]: chan_sip.c:3080 sip_destroy: Destroying SIP dialog 6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35 [Jan 10 17:56:37] Really destroying SIP dialog '6a741e0e2c8e7daf3b528dcb3e106e82@84.234.24.35' Method: INVITE [Jan 10 17:56:39] <--- SIP read from 83.105.95.134:43460 ---> INVITE sip:1003@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK98f7509ec9c80a40 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: Contact: Supported: replaces, timer Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33197 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8001 8000 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1003@www.jb-consultancy.com SIP/2.0 (46) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK98f7509ec9c80a40 (63) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (37) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces, timer (26) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: CSeq: 33197 INVITE (18) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Type: application/sdp (29) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 331 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: (0) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8001 8000 IN IP4 10.0.0.213 (34) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:39] --- (13 headers 16 lines) --- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to Off [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for 0ab7060411db3108@10.0.0.213 - INVITE (With RTP) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1662 parse_sip_options: Begin: parsing SIP "Supported: replaces, timer" [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1670 parse_sip_options: Found SIP option: -replaces- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1676 parse_sip_options: Matched SIP option: replaces [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1670 parse_sip_options: Found SIP option: -timer- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1676 parse_sip_options: Matched SIP option: timer [Jan 10 17:56:39] Sending to 10.0.0.213 : 3876 (no NAT) [Jan 10 17:56:39] Using INVITE request as basis request - 0ab7060411db3108@10.0.0.213 [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:39] <--- Reliably Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK98f7509ec9c80a40;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as11264af4 Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33197 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41f279ee" Content-Length: 0 <------------> [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #129 [Jan 10 17:56:39] Scheduling destruction of SIP dialog '0ab7060411db3108@10.0.0.213' in 32000 ms (Method: INVITE) [Jan 10 17:56:39] Found user '1001' [Jan 10 17:56:39] <--- SIP read from 83.105.95.134:43460 ---> ACK sip:1003@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK98f7509ec9c80a40 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as11264af4 Contact: Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33197 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:1003@www.jb-consultancy.com SIP/2.0 (43) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK98f7509ec9c80a40 (63) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as11264af4 (52) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 33197 ACK (15) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (21) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:39] --- (11 headers 0 lines) --- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #129 [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0ab7060411db3108@10.0.0.213' of Response 33197: Match Not Found [Jan 10 17:56:39] <--- SIP read from 83.105.95.134:43460 ---> INVITE sip:1003@www.jb-consultancy.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: Contact: Supported: replaces, timer Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@www.jb-consultancy.com", nonce="41f279ee", response="ead56631c146f844d9307b2023744891" Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8001 8001 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1003@www.jb-consultancy.com SIP/2.0 (46) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7 (63) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (37) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces, timer (26) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@www.jb-consultancy.com", nonce="41f279ee", response="ead56631c146f844d9307b2023744891" (178) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: CSeq: 33198 INVITE (18) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 331 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8001 8001 IN IP4 10.0.0.213 (34) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:39] --- (14 headers 16 lines) --- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:39] Sending to 83.105.95.134 : 43460 (NAT) [Jan 10 17:56:39] Using INVITE request as basis request - 0ab7060411db3108@10.0.0.213 [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:39] Found user '1001' [Jan 10 17:56:39] Found RTP audio format 0 [Jan 10 17:56:39] Found RTP audio format 3 [Jan 10 17:56:39] Found RTP audio format 4 [Jan 10 17:56:39] Found RTP audio format 8 [Jan 10 17:56:39] Found RTP audio format 18 [Jan 10 17:56:39] Found RTP audio format 2 [Jan 10 17:56:39] Found RTP audio format 101 [Jan 10 17:56:39] Peer audio RTP is at port 10.0.0.213:5956 [Jan 10 17:56:39] Found description format PCMU for ID 0 [Jan 10 17:56:39] Found description format GSM for ID 3 [Jan 10 17:56:39] Found description format G723 for ID 4 [Jan 10 17:56:39] Found description format PCMA for ID 8 [Jan 10 17:56:39] Found description format G729 for ID 18 [Jan 10 17:56:39] Found description format G726-32 for ID 2 [Jan 10 17:56:39] Found description format telephone-event for ID 101 [Jan 10 17:56:39] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel [Jan 10 17:56:39] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:39] Peer audio RTP is at port 10.0.0.213:5956 [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:13237 handle_request_invite: Checking SIP call limits for device 1001 [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3048 update_call_counter: Call from peer '1001' is 2 out of 200 [Jan 10 17:56:39] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:39] Looking for 1003 in int_jbc (domain www.jb-consultancy.com) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:7891 build_route: build_route: Contact hop: [Jan 10 17:56:39] list_route: hop: [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:13310 handle_request_invite: SIP/1001-081bd2a8: New call is still down.... Trying... [Jan 10 17:56:39] <--- Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:39] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081bd2a8 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:39] DEBUG[14596]: pbx.c:1767 pbx_extension_helper: Launching 'Set' [Jan 10 17:56:39] -- Executing [1003@int_jbc:1] Set("SIP/1001-081bd2a8", "OUTTRUNK=JBC_OUT") in new stack [Jan 10 17:56:39] DEBUG[14596]: pbx.c:1767 pbx_extension_helper: Launching 'NoOp' [Jan 10 17:56:39] -- Executing [1003@int_jbc:2] NoOp("SIP/1001-081bd2a8", "") in new stack [Jan 10 17:56:39] DEBUG[14596]: pbx.c:1767 pbx_extension_helper: Launching 'Dial' [Jan 10 17:56:39] -- Executing [1003@int_jbc:3] Dial("SIP/1001-081bd2a8", "SIP/1003|30|tT") in new stack [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:15106 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:2559 do_setnat: Setting NAT on RTP to On [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3766 sip_new: *** Our native formats are 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3767 sip_new: *** Joint capabilities are 0x0 (nothing) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3768 sip_new: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3769 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3771 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3792 sip_new: This channel will not be able to handle video. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1003-3. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1003-2. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable OUTTRUNK. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable STACK-int_jbc-1003-1. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Jan 10 17:56:39] DEBUG[14596]: channel.c:3185 ast_channel_inherit_variables: Not copying variable SIPURI. [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:2812 sip_call: Outgoing Call for 1003 [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:3048 update_call_counter: Call to peer '1003' is 1 out of 200 [Jan 10 17:56:39] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:2826 sip_call: Our T38 capability (0), joint T38 capability (0) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: False [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Jan 10 17:56:39] Audio is at 84.234.24.35 port 25714 [Jan 10 17:56:39] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:39] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:39] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:39] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 (70) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport (63) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=as0a5146c7 (65) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 3: To: (61) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 4: Contact: (32) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 (54) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:39 GMT (35) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 287 (19) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: o=root 14596 14596 IN IP4 84.234.24.35 (38) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: s=session (9) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: c=IN IP4 84.234.24.35 (21) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: m=audio 25714 RTP/AVP 0 8 3 101 (31) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-16 (15) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=silenceSupp:off - - - - (25) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:39] Reliably Transmitting (NAT) to 83.105.95.134:44780: INVITE sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport From: "James B Warrington" ;tag=as0a5146c7 To: Contact: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 287 v=0 o=root 14596 14596 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 25714 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jan 10 17:56:39] DEBUG[14596]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #131 [Jan 10 17:56:39] -- Called 1003 [Jan 10 17:56:39] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 Contact: To: ;tag=da43787a From: "James B Warrington";tag=as0a5146c7 Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 INVITE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 (68) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=da43787a (74) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "James B Warrington";tag=as0a5146c7 (64) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 (54) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: (0) [Jan 10 17:56:39] --- (9 headers 0 lines) --- [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2104 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #131 - INVITE (got response) [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:2113 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' Request 102: Found [Jan 10 17:56:39] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 180 to standard invite [Jan 10 17:56:39] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081d4f18 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 6 (Ringing) [Jan 10 17:56:39] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:39] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:39] -- SIP/1003-081d4f18 is ringing [Jan 10 17:56:39] <--- Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:41] DEBUG[14594]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Jan 10 17:56:42] <--- SIP read from 90.195.58.25:50518 ---> <-------------> [Jan 10 17:56:42] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: (0) [Jan 10 17:56:42] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: (0) [Jan 10 17:56:42] --- (0 headers 1 lines) --- [Jan 10 17:56:45] DEBUG[14596]: rtp.c:857 ast_rtcp_read: RTCP NAT: Got RTCP from other end. Now sending to address 83.105.95.134:34083 [Jan 10 17:56:45] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 132 bytes [Jan 10 17:56:45] DEBUG[14596]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:34082 [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 10 17:56:45] Audio is at 84.234.24.35 port 17624 [Jan 10 17:56:45] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:45] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:45] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:45] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] <--- Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 14596 14596 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 17624 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 10 17:56:45] DEBUG[14596]: rtp.c:2647 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 10 17:56:45] DEBUG[14596]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:45] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 Contact: To: ;tag=da43787a From: "James B Warrington";tag=as0a5146c7 Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 183 v=0 o=- 9 2 IN IP4 10.0.0.250 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.250 t=0 0 m=audio 26082 RTP/AVP 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 (68) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=da43787a (74) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "James B Warrington";tag=as0a5146c7 (64) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 (54) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 183 (19) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=- 9 2 IN IP4 10.0.0.250 (25) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.250 (19) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 26082 RTP/AVP 0 8 3 101 (31) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:45] --- (11 headers 9 lines) --- [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2053 __sip_ack: Acked pending invite 102 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:11531 handle_response_invite: SIP response 200 to standard invite [Jan 10 17:56:45] Found RTP audio format 0 [Jan 10 17:56:45] Found RTP audio format 8 [Jan 10 17:56:45] Found RTP audio format 3 [Jan 10 17:56:45] Found RTP audio format 101 [Jan 10 17:56:45] Peer audio RTP is at port 10.0.0.250:26082 [Jan 10 17:56:45] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:45] Found description format telephone-event for ID 101 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1003-081d4f18 [Jan 10 17:56:45] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:45] Peer audio RTP is at port 10.0.0.250:26082 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:45] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 2 (In use) [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:7891 build_route: build_route: Contact hop: [Jan 10 17:56:45] list_route: hop: [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5576 reqprep: Strict routing enforced for session 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 [Jan 10 17:56:45] set_destination: Parsing for address/port to send to [Jan 10 17:56:45] set_destination: set destination to 83.105.95.134, port 44780 [Jan 10 17:56:45] Transmitting (NAT) to 83.105.95.134:44780: ACK sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK18403d0f;rport From: "James B Warrington" ;tag=as0a5146c7 To: ;tag=da43787a Contact: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:45] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081d4f18 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 2 (In use) [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] -- SIP/1003-081d4f18 answered SIP/1001-081bd2a8 [Jan 10 17:56:45] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081bd2a8 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:3428 sip_answer: SIP answering channel: SIP/1001-081bd2a8 [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 10 17:56:45] Audio is at 84.234.24.35 port 17624 [Jan 10 17:56:45] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:45] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:45] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:45] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] <--- Reliably Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKf628efea765acce7;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 14596 14597 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 17624 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 10 17:56:45] DEBUG[14596]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #135 [Jan 10 17:56:45] DEBUG[14596]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:34082 [Jan 10 17:56:45] DEBUG[14596]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:45] DEBUG[14596]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:45540 [Jan 10 17:56:45] DEBUG[14596]: rtp.c:2647 ast_rtp_write: Ooh, format changed from unknown to ulaw [Jan 10 17:56:45] DEBUG[14596]: rtp.c:2664 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Jan 10 17:56:45] <--- SIP read from 83.105.95.134:43460 ---> ACK sip:1003@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK95ef8155ae50a35f From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Contact: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="1b4bb85759f9c9acd6e4cf207a1219ea" Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33198 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:1003@84.234.24.35 SIP/2.0 (33) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK95ef8155ae50a35f (63) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as6ffc6d9b (52) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="1b4bb85759f9c9acd6e4cf207a1219ea" (168) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: CSeq: 33198 ACK (15) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (21) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:45] --- (12 headers 0 lines) --- [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #135 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0ab7060411db3108@10.0.0.213' of Response 33198: Match Not Found [Jan 10 17:56:45] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 Contact: To: ;tag=da43787a From: "James B Warrington";tag=as0a5146c7 Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 183 v=0 o=- 9 2 IN IP4 10.0.0.250 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.0.250 t=0 0 m=audio 26082 RTP/AVP 0 8 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0b7a2452;rport=5060 (68) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=da43787a (74) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "James B Warrington";tag=as0a5146c7 (64) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 (54) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 INVITE (16) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Type: application/sdp (29) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 183 (19) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=- 9 2 IN IP4 10.0.0.250 (25) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=CounterPath X-Lite 3.0 (24) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.250 (19) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 26082 RTP/AVP 0 8 3 101 (31) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-15 (15) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendrecv (10) [Jan 10 17:56:45] --- (11 headers 9 lines) --- [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' of Request 102: Match Found [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:11529 handle_response_invite: SIP response 200 to RE-invite on outgoing call 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 [Jan 10 17:56:45] Found RTP audio format 0 [Jan 10 17:56:45] Found RTP audio format 8 [Jan 10 17:56:45] Found RTP audio format 3 [Jan 10 17:56:45] Found RTP audio format 101 [Jan 10 17:56:45] Peer audio RTP is at port 10.0.0.250:26082 [Jan 10 17:56:45] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:45] Found description format telephone-event for ID 101 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1003-081d4f18 [Jan 10 17:56:45] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:45] Peer audio RTP is at port 10.0.0.250:26082 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:45] DEBUG[14576]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 2 (In use) [Jan 10 17:56:45] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:45] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:7830 build_route: build_route: Retaining previous route: [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:11662 handle_response_invite: Strange... The other side of the bridge does not have a udptl struct [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:11667 handle_response_invite: T38 state changed to 0 on channel SIP [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:11670 handle_response_invite: T38 state changed to 0 on channel SIP/1003-081d4f18 [Jan 10 17:56:45] DEBUG[14576]: chan_sip.c:5576 reqprep: Strict routing enforced for session 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 [Jan 10 17:56:45] set_destination: Parsing for address/port to send to [Jan 10 17:56:45] set_destination: set destination to 83.105.95.134, port 44780 [Jan 10 17:56:45] Transmitting (NAT) to 83.105.95.134:44780: ACK sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK686fe9c7;rport From: "James B Warrington" ;tag=as0a5146c7 To: ;tag=da43787a Contact: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:45] DEBUG[14596]: rtp.c:1140 ast_rtp_read: RTP NAT: Got audio from other end. Now sending to address 83.105.95.134:34082 [Jan 10 17:56:46] DEBUG[14594]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Jan 10 17:56:46] <--- SIP read from 83.105.95.134:44780 ---> <-------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: (0) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: (0) [Jan 10 17:56:46] --- (0 headers 1 lines) --- [Jan 10 17:56:46] <--- SIP read from 83.105.95.134:43460 ---> INVITE sip:1003@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK0bbfaea2236c8f84 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Contact: Supported: replaces, timer Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="ad21ab754988886f06c31d10a6784613" Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33199 INVITE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 331 v=0 o=1001 8001 8002 IN IP4 10.0.0.213 s=SIP Call c=IN IP4 10.0.0.213 t=0 0 m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: INVITE sip:1003@84.234.24.35 SIP/2.0 (36) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK0bbfaea2236c8f84 (63) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as6ffc6d9b (52) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces, timer (26) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="ad21ab754988886f06c31d10a6784613" (168) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: CSeq: 33199 INVITE (18) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 70 (16) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Type: application/sdp (29) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 331 (19) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: v=0 (3) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: o=1001 8001 8002 IN IP4 10.0.0.213 (34) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: s=SIP Call (10) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: c=IN IP4 10.0.0.213 (19) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: t=0 0 (5) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: m=audio 5956 RTP/AVP 0 3 4 8 18 2 101 (37) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=sendonly (10) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=ptime:20 (10) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4559 parse_request: Line: a=fmtp:101 0-11 (15) [Jan 10 17:56:46] --- (14 headers 16 lines) --- [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Jan 10 17:56:46] Sending to 83.105.95.134 : 43460 (NAT) [Jan 10 17:56:46] Found RTP audio format 0 [Jan 10 17:56:46] Found RTP audio format 3 [Jan 10 17:56:46] Found RTP audio format 4 [Jan 10 17:56:46] Found RTP audio format 8 [Jan 10 17:56:46] Found RTP audio format 18 [Jan 10 17:56:46] Found RTP audio format 2 [Jan 10 17:56:46] Found RTP audio format 101 [Jan 10 17:56:46] Peer audio RTP is at port 10.0.0.213:5956 [Jan 10 17:56:46] Found description format PCMU for ID 0 [Jan 10 17:56:46] Found description format GSM for ID 3 [Jan 10 17:56:46] Found description format G723 for ID 4 [Jan 10 17:56:46] Found description format PCMA for ID 8 [Jan 10 17:56:46] Found description format G729 for ID 18 [Jan 10 17:56:46] Found description format G726-32 for ID 2 [Jan 10 17:56:46] Found description format telephone-event for ID 101 [Jan 10 17:56:46] Got unsupported a:fmtp in SDP offer [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:5068 process_sdp: T38 state changed to 0 on channel SIP/1001-081bd2a8 [Jan 10 17:56:46] Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 10 17:56:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 10 17:56:46] Peer audio RTP is at port 10.0.0.213:5956 [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:5145 process_sdp: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:5152 process_sdp: We have an owner, now see if we need to change this call [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:13288 handle_request_invite: Got a SIP re-invite for call 0ab7060411db3108@10.0.0.213 [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:13383 handle_request_invite: SIP/1001-081bd2a8: This call is UP.... [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:6354 transmit_response_with_sdp: Setting framing from config on incoming call [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:6122 add_sdp: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:6123 add_sdp: ** Our prefcodec: 0x0 (nothing) [Jan 10 17:56:46] Audio is at 84.234.24.35 port 17624 [Jan 10 17:56:46] Adding codec 0x4 (ulaw) to SDP [Jan 10 17:56:46] Adding codec 0x8 (alaw) to SDP [Jan 10 17:56:46] Adding codec 0x2 (gsm) to SDP [Jan 10 17:56:46] Adding non-codec 0x1 (telephone-event) to SDP [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:6254 add_sdp: -- Done with adding codecs to SDP [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:6299 add_sdp: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 10 17:56:46] <--- Reliably Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK0bbfaea2236c8f84;received=83.105.95.134 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33199 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 14596 14598 IN IP4 84.234.24.35 s=session c=IN IP4 84.234.24.35 t=0 0 m=audio 17624 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #136 [Jan 10 17:56:46] -- Started music on hold, class 'default', on SIP/1003-081d4f18 [Jan 10 17:56:46] DEBUG[14596]: channel.c:1908 ast_settimeout: Scheduling timer at 160 sample intervals [Jan 10 17:56:46] DEBUG[14596]: channel.c:2225 __ast_read: Generator got voice, switching to phase locked mode [Jan 10 17:56:46] DEBUG[14596]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 10 17:56:46] DEBUG[14596]: channel.c:2731 set_format: Set channel SIP/1003-081d4f18 to write format slin [Jan 10 17:56:46] DEBUG[14596]: res_musiconhold.c:250 ast_moh_files_next: SIP/1003-081d4f18 Opened file 1 '/var/lib/asterisk/moh/fpm-sunshine' [Jan 10 17:56:46] <--- SIP read from 83.105.95.134:43460 ---> ACK sip:1003@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK2d41a042cb2f7658 From: "James B Warrington" ;tag=de5d0225e43ffba6 To: ;tag=as6ffc6d9b Contact: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="1b4bb85759f9c9acd6e4cf207a1219ea" Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 33199 ACK User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: ACK sip:1003@84.234.24.35 SIP/2.0 (33) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK2d41a042cb2f7658 (63) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James B Warrington" ;tag=de5d0225e43ffba6 (81) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=as6ffc6d9b (52) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:1003@84.234.24.35", nonce="41f279ee", response="1b4bb85759f9c9acd6e4cf207a1219ea" (168) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: CSeq: 33199 ACK (15) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Max-Forwards: 70 (16) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (21) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:46] --- (12 headers 0 lines) --- [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received ACK (6) - Command in SIP ACK [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #136 [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0ab7060411db3108@10.0.0.213' of Response 33199: Match Not Found [Jan 10 17:56:46] <--- SIP read from 89.213.46.56:64435 ---> NOTIFY sip:84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b9f9908c From: "James Brindle" ;tag=16273a752963b43co3 To: Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1823 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: NOTIFY sip:84.234.24.35 SIP/2.0 (31) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b9f9908c (59) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James Brindle" ;tag=16273a752963b43co3 (68) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (22) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 448cad21-a7370180@192.168.1.211 (40) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1823 NOTIFY (17) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Event: keep-alive (17) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Linksys/SPA941-4.1.15 (33) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:56:46] --- (10 headers 0 lines) --- [Jan 10 17:56:46] <--- Transmitting (no NAT) to 89.213.46.56:64435 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-b9f9908c;received=89.213.46.56 From: "James Brindle" ;tag=16273a752963b43co3 To: ;tag=as485c1aeb Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1823 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 342 [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4264 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: OPTIONS sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 (71) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK11148c65;rport (63) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "asterisk" ;tag=as0f8fe4b8 (59) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (61) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (36) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 5522038701aacd963542dac443b6c123@84.234.24.35 (54) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Max-Forwards: 70 (16) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Date: Wed, 10 Jan 2007 17:56:46 GMT (35) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Supported: replaces (19) [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Content-Length: 0 (17) [Jan 10 17:56:46] Reliably Transmitting (NAT) to 83.105.95.134:44780: OPTIONS sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK11148c65;rport From: "asterisk" ;tag=as0f8fe4b8 To: Contact: Call-ID: 5522038701aacd963542dac443b6c123@84.234.24.35 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 17:56:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jan 10 17:56:46] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #137 [Jan 10 17:56:47] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK11148c65;rport=5060 Contact: To: ;tag=4809be2e From: "asterisk";tag=as0f8fe4b8 Call-ID: 5522038701aacd963542dac443b6c123@84.234.24.35 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 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK11148c65;rport=5060 (68) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (31) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=4809be2e (74) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "asterisk";tag=as0f8fe4b8 (58) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 5522038701aacd963542dac443b6c123@84.234.24.35 (54) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Accept: application/sdp (23) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Accept-Language: en (19) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:47] --- (12 headers 0 lines) --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #137 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '5522038701aacd963542dac443b6c123@84.234.24.35' of Request 102: Match Not Found [Jan 10 17:56:47] Really destroying SIP dialog '5522038701aacd963542dac443b6c123@84.234.24.35' Method: OPTIONS [Jan 10 17:56:47] <--- SIP read from 83.105.95.134:43460 ---> REFER sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK4659771193cb40d3 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Contact: Supported: replaces Refer-To: Referred-By: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64437 REFER User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: REFER sip:907870920797@84.234.24.35 SIP/2.0 (43) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK4659771193cb40d3 (63) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Contact: (35) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Supported: replaces (19) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Refer-To: (134) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Referred-By: (46) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: CSeq: 64437 REFER (17) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Max-Forwards: 70 (16) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 13: Content-Length: 0 (21) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 14: (0) [Jan 10 17:56:47] --- (14 headers 0 lines) --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received REFER (9) - Command in SIP REFER [Jan 10 17:56:47] Call 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 got a SIP call transfer from caller: (REFER)! [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:8623 get_refer_info: Attended transfer: Will use Replace-Call-ID : 0ab7060411db3108@10.0.0.213 (No check of from/to tags) [Jan 10 17:56:47] SIP transfer to extension 1003@int_jbc by 1001@www.jb-consultancy.com [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:13794 handle_request_refer: SIP attended transfer: Transferer channel SIP/1001-081d0fb0, transferee channel SIP/7169832-081bec40 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:13810 handle_request_refer: Got SIP transfer, applying to bridged peer 'SIP/7169832-081bec40' [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:8477 get_sip_pvt_byid_locked: Looking for callid 0ab7060411db3108@10.0.0.213 (fromtag de5d0225e43ffba6 totag as6ffc6d9b) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:8501 get_sip_pvt_byid_locked: Matched INCOMING call - their tag is de5d0225e43ffba6 Our tag is as6ffc6d9b [Jan 10 17:56:47] <--- Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bK4659771193cb40d3;received=83.105.95.134 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64437 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:13565 local_attended_transfer: SIP attended transfer: trying to bridge SIP/1001-081bd2a8 and SIP/7169832-081bec40 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12579 attempt_transfer: Sip transfer:-------------------- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12581 attempt_transfer: -- Transferer to PBX channel: SIP/1001-081d0fb0 State Up [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12585 attempt_transfer: -- Transferer to PBX second channel (target): SIP/1001-081bd2a8 State Up [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12589 attempt_transfer: -- Bridged call to transferee: SIP/7169832-081bec40 State Up [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12593 attempt_transfer: -- Bridged call to transfer target: SIP/1003-081d4f18 State Up [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12596 attempt_transfer: -- END Sip transfer:-------------------- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12604 attempt_transfer: SIP transfer: Four channels to handle [Jan 10 17:56:47] -- Stopped music on hold on SIP/7169832-081bec40 [Jan 10 17:56:47] DEBUG[14576]: channel.c:2731 set_format: Set channel SIP/7169832-081bec40 to write format ulaw [Jan 10 17:56:47] DEBUG[14576]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 10 17:56:47] -- Stopped music on hold on SIP/1003-081d4f18 [Jan 10 17:56:47] DEBUG[14576]: channel.c:2731 set_format: Set channel SIP/1003-081d4f18 to write format ulaw [Jan 10 17:56:47] DEBUG[14576]: channel.c:1908 ast_settimeout: Scheduling timer at 0 sample intervals [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12635 attempt_transfer: SIP transfer: trying to masquerade SIP/7169832-081bec40 into SIP/1001-081bd2a8 [Jan 10 17:56:47] DEBUG[14576]: channel.c:3119 ast_channel_masquerade: Planning to masquerade channel SIP/7169832-081bec40 into the structure of SIP/1001-081bd2a8 [Jan 10 17:56:47] DEBUG[14576]: channel.c:3132 ast_channel_masquerade: Done planning to masquerade channel SIP/7169832-081bec40 into the structure of SIP/1001-081bd2a8 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:12640 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:5576 reqprep: Strict routing enforced for session 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:47] set_destination: Parsing for address/port to send to [Jan 10 17:56:47] set_destination: set destination to 10.0.0.213, port 3876 [Jan 10 17:56:47] Reliably Transmitting (NAT) to 83.105.95.134:43460: NOTIFY sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK584cec1e;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Contact: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=64437 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #140 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:13593 local_attended_transfer: SIP attended transfer: Unlocking channel SIP/1001-081bd2a8 [Jan 10 17:56:47] DEBUG[14596]: channel.c:3249 ast_do_masquerade: Actually Masquerading SIP/7169832-081bec40(6) into the structure of SIP/1001-081bd2a8(6) [Jan 10 17:56:47] DEBUG[14596]: channel.c:3261 ast_do_masquerade: Got clone lock for masquerade on 'SIP/7169832-081bec40' at 0x81ba598 [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:3553 sip_fixup: SIP Fixup: New owner for dialogue 0ab7060411db3108@10.0.0.213: SIP/7169832-081bec40 (Old parent: SIP/7169832-081bec40) [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:3278 sip_hangup: Hangup call SIP/7169832-081bec40, SIP callid 0ab7060411db3108@10.0.0.213) [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:3286 sip_hangup: update_call_counter(1001) - decrement call limit counter on hangup [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:3022 update_call_counter: Call from peer '1001' removed from call limit 200 [Jan 10 17:56:47] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] Scheduling destruction of SIP dialog '0ab7060411db3108@10.0.0.213' in 32000 ms (Method: ACK) [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:5576 reqprep: Strict routing enforced for session 0ab7060411db3108@10.0.0.213 [Jan 10 17:56:47] set_destination: Parsing for address/port to send to [Jan 10 17:56:47] set_destination: set destination to 10.0.0.213, port 3876 [Jan 10 17:56:47] Reliably Transmitting (NAT) to 83.105.95.134:43460: BYE sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK763f4bcd;rport From: ;tag=as6ffc6d9b To: "James B Warrington" ;tag=de5d0225e43ffba6 Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #142 [Jan 10 17:56:47] DEBUG[14596]: channel.c:3456 ast_do_masquerade: Putting channel SIP/7169832-081bec40 in 4/4 formats [Jan 10 17:56:47] DEBUG[14596]: chan_sip.c:3553 sip_fixup: SIP Fixup: New owner for dialogue 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185: SIP/7169832-081bec40 (Old parent: SIP/1001-081bd2a8) [Jan 10 17:56:47] DEBUG[14596]: channel.c:3491 ast_do_masquerade: Released clone lock on 'SIP/1001-081bd2a8' [Jan 10 17:56:47] DEBUG[14596]: channel.c:3501 ast_do_masquerade: Done Masquerading SIP/7169832-081bec40 (6) [Jan 10 17:56:47] DEBUG[14594]: channel.c:3896 ast_channel_bridge: Bridge stops because we're zombie or need a soft hangup: c0=SIP/1001-081bd2a8, c1=SIP/1001-081d0fb0, flags: Yes,Yes,No,No [Jan 10 17:56:47] DEBUG[14594]: channel.c:3990 ast_channel_bridge: Bridge stops bridging channels SIP/1001-081bd2a8 and SIP/1001-081d0fb0 [Jan 10 17:56:47] DEBUG[14594]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/1001-081d0fb0' [Jan 10 17:56:47] DEBUG[14594]: chan_sip.c:3263 sip_hangup: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35. [Jan 10 17:56:47] Scheduling destruction of SIP dialog '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' in 6400 ms (Method: REFER) [Jan 10 17:56:47] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081d0fb0 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] DEBUG[14594]: rtp.c:1465 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 10 17:56:47] DEBUG[14594]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 10 17:56:47] DEBUG[14594]: pbx.c:2363 __ast_pbx_run: Spawn extension (incoming,8001,4) exited non-zero on 'SIP/1001-081bd2a8' [Jan 10 17:56:47] == Spawn extension (incoming, 8001, 4) exited non-zero on 'SIP/1001-081bd2a8' [Jan 10 17:56:47] DEBUG[14594]: channel.c:1611 ast_hangup: Hanging up zombie 'SIP/1001-081bd2a8' [Jan 10 17:56:47] DEBUG[14594]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1001-081bd2a8 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1001 - state 2 (In use) [Jan 10 17:56:47] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1001 [Jan 10 17:56:47] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1001 [Jan 10 17:56:47] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK584cec1e;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK584cec1e;rport (63) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=f37978bef01506b5 (51) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 103 NOTIFY (16) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (35) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Supported: replaces, timer (26) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:47] --- (11 headers 0 lines) --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #140 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' of Request 103: Match Not Found [Jan 10 17:56:47] SIP Response message for INCOMING dialog NOTIFY arrived [Jan 10 17:56:47] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK763f4bcd;rport From: ;tag=as6ffc6d9b To: "James B Warrington" ;tag=de5d0225e43ffba6 Call-ID: 0ab7060411db3108@10.0.0.213 CSeq: 102 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer Content-Length: 0 <-------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK763f4bcd;rport (63) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=as6ffc6d9b (54) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "James B Warrington" ;tag=de5d0225e43ffba6 (79) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 0ab7060411db3108@10.0.0.213 (36) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 102 BYE (13) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Contact: (35) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Supported: replaces, timer (26) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Content-Length: 0 (17) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: (0) [Jan 10 17:56:47] --- (11 headers 0 lines) --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #142 [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '0ab7060411db3108@10.0.0.213' of Request 102: Match Not Found [Jan 10 17:56:47] SIP Response message for INCOMING dialog BYE arrived [Jan 10 17:56:47] Really destroying SIP dialog '0ab7060411db3108@10.0.0.213' Method: ACK [Jan 10 17:56:47] <--- SIP read from 83.105.95.134:43460 ---> BYE sip:907870920797@84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKc65e28d7f6eadf64 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64438 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: BYE sip:907870920797@84.234.24.35 SIP/2.0 (41) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKc65e28d7f6eadf64 (63) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: ;tag=f37978bef01506b5 (53) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: "JBC: 07870920797" ;tag=as66e05052 (69) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 64438 BYE (15) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Max-Forwards: 70 (16) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE (85) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (21) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:56:47] --- (10 headers 0 lines) --- [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received BYE (8) - Command in SIP BYE [Jan 10 17:56:47] Sending to 83.105.95.134 : 43460 (NAT) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:1615 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:47] Scheduling destruction of SIP dialog '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' in 6400 ms (Method: BYE) [Jan 10 17:56:47] DEBUG[14576]: chan_sip.c:14034 handle_request_bye: Received bye, no owner, selfdestruct soon. .[Jan 10 17:56:47] <--- Transmitting (NAT) to 83.105.95.134:43460 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.213:3876;branch=z9hG4bKc65e28d7f6eadf64;received=83.105.95.134 From: ;tag=f37978bef01506b5 To: "JBC: 07870920797" ;tag=as66e05052 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 64438 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:48] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 10 17:56:51] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Jan 10 17:56:51] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:1992 __sip_autodestruct: Auto destroying SIP dialog '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:1999 __sip_autodestruct: Finally hanging up channel after transfer: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:5576 reqprep: Strict routing enforced for session 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 [Jan 10 17:56:54] set_destination: Parsing for address/port to send to [Jan 10 17:56:54] set_destination: set destination to 10.0.0.213, port 3876 [Jan 10 17:56:54] Reliably Transmitting (NAT) to 83.105.95.134:43460: BYE sip:1001@10.0.0.213:3876 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7bb76d7a;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #145 [Jan 10 17:56:54] Scheduling destruction of SIP dialog '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' in 6400 ms (Method: BYE) [Jan 10 17:56:54] <--- SIP read from 83.105.95.134:43460 ---> SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7bb76d7a;rport From: "JBC: 07870920797" ;tag=as66e05052 To: ;tag=f37978bef01506b5 Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 CSeq: 104 BYE User-Agent: Grandstream GXP2000 1.1.1.14 Content-Length: 0 <-------------> [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 481 No Such Call (24) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK7bb76d7a;rport (63) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "JBC: 07870920797" ;tag=as66e05052 (71) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=f37978bef01506b5 (51) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35 (54) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 104 BYE (13) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: User-Agent: Grandstream GXP2000 1.1.1.14 (40) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Content-Length: 0 (17) [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: (0) [Jan 10 17:56:54] --- (8 headers 0 lines) --- [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #145 [Jan 10 17:56:54] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' of Request 104: Match Not Found [Jan 10 17:56:54] SIP Response message for INCOMING dialog BYE arrived [Jan 10 17:56:54] Really destroying SIP dialog '4d58ff4f663d6f7f0b5a5a4a74c7339a@84.234.24.35' Method: BYE [Jan 10 17:56:54] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 176 bytes [Jan 10 17:56:56] DEBUG[14596]: rtp.c:862 ast_rtcp_read: Got RTCP report of 64 bytes [Jan 10 17:56:56] <--- SIP read from 217.14.132.178:5060 ---> BYE sip:8001@84.234.24.35 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK3dc5.cd4f0d05.0 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK0472c384;rport=5060 From: "07870920797" ;tag=as03fe6293 To: ;tag=as1e337580 Contact: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 103 BYE User-Agent: Telappliant VoIP Gateway Max-Forwards: 16 Content-Length: 0 <-------------> [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: BYE sip:8001@84.234.24.35 SIP/2.0 (33) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Record-Route: (56) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK3dc5.cd4f0d05.0 (61) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK0472c384;rport=5060 (69) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "07870920797" ;tag=as03fe6293 (67) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: To: ;tag=as1e337580 (51) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Contact: (40) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 (56) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: CSeq: 103 BYE (13) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: User-Agent: Telappliant VoIP Gateway (36) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: Max-Forwards: 16 (16) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 11: Content-Length: 0 (17) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 12: (0) [Jan 10 17:56:56] --- (12 headers 0 lines) --- [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:14453 handle_request: **** Received BYE (8) - Command in SIP BYE [Jan 10 17:56:56] Sending to 217.14.132.178 : 5060 (no NAT) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:1615 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:14030 handle_request_bye: Received bye, issuing owner hangup .[Jan 10 17:56:56] <--- Transmitting (no NAT) to 217.14.132.178:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 217.14.132.178;branch=z9hG4bK3dc5.cd4f0d05.0;received=217.14.132.178 Via: SIP/2.0/UDP 217.14.138.48:5060;branch=z9hG4bK0472c384;rport=5060 Record-Route: From: "07870920797" ;tag=as03fe6293 To: ;tag=as1e337580 Call-ID: 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 10 17:56:56] DEBUG[14596]: channel.c:3682 ast_generic_bridge: Didn't get a frame from channel: SIP/7169832-081bec40 [Jan 10 17:56:56] DEBUG[14596]: channel.c:3990 ast_channel_bridge: Bridge stops bridging channels SIP/7169832-081bec40 and SIP/1003-081d4f18 [Jan 10 17:56:56] DEBUG[14596]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/1003-081d4f18' [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:3278 sip_hangup: Hangup call SIP/1003-081d4f18, SIP callid 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35) [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:3286 sip_hangup: update_call_counter(1003) - decrement call limit counter on hangup [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:2978 update_call_counter: Updating call counter for outgoing call [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:3022 update_call_counter: Call to peer '1003' removed from call limit 200 [Jan 10 17:56:56] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003 [Jan 10 17:56:56] Scheduling destruction of SIP dialog '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' in 6400 ms (Method: INVITE) [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:5576 reqprep: Strict routing enforced for session 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 [Jan 10 17:56:56] set_destination: Parsing for address/port to send to [Jan 10 17:56:56] set_destination: set destination to 83.105.95.134, port 44780 [Jan 10 17:56:56] Reliably Transmitting (NAT) to 83.105.95.134:44780: BYE sip:1003@83.105.95.134:44780;rinstance=f17893ec71eac0e2 SIP/2.0 Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0cf2d02a;rport From: "James B Warrington" ;tag=as0a5146c7 To: ;tag=da43787a Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:1957 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #148 [Jan 10 17:56:56] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/1003-081d4f18 [Jan 10 17:56:56] DEBUG[14596]: rtp.c:1465 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Jan 10 17:56:56] DEBUG[14596]: app_dial.c:1643 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jan 10 17:56:56] DEBUG[14596]: pbx.c:2363 __ast_pbx_run: Spawn extension (int_jbc,1003,3) exited non-zero on 'SIP/7169832-081bec40' [Jan 10 17:56:56] == Spawn extension (int_jbc, 1003, 3) exited non-zero on 'SIP/7169832-081bec40' [Jan 10 17:56:56] DEBUG[14596]: channel.c:1606 ast_hangup: Hanging up channel 'SIP/7169832-081bec40' [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:3278 sip_hangup: Hangup call SIP/7169832-081bec40, SIP callid 7e69b30d05593e6a1482ff6a404075e5@217.14.132.185) [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:3286 sip_hangup: update_call_counter(7169832) - decrement call limit counter on hangup [Jan 10 17:56:56] DEBUG[14596]: chan_sip.c:2978 update_call_counter: Updating call counter for incoming call [Jan 10 17:56:56] DEBUG[14596]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/7169832-081bec40 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:56] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:56] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:56] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/1003 - state 1 (Not in use) [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 1003 [Jan 10 17:56:56] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 1003 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 7169832 [Jan 10 17:56:56] DEBUG[14573]: chan_sip.c:15048 sip_devicestate: Checking device state for peer 7169832 [Jan 10 17:56:56] DEBUG[14573]: devicestate.c:287 do_state_change: Changing state for SIP/7169832 - state 4 (Invalid) [Jan 10 17:56:56] <--- SIP read from 83.105.95.134:44780 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0cf2d02a;rport=5060 Contact: To: ;tag=da43787a From: "James B Warrington";tag=as0a5146c7 Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 CSeq: 103 BYE User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 <-------------> [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: SIP/2.0 200 OK (14) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 84.234.24.35:5060;branch=z9hG4bK0cf2d02a;rport=5060 (68) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: Contact: (66) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: ;tag=da43787a (74) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: From: "James B Warrington";tag=as0a5146c7 (64) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: Call-ID: 6fd78ff64582f95c3deec38607fdfd19@84.234.24.35 (54) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: CSeq: 103 BYE (13) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: User-Agent: X-Lite release 1006e stamp 34025 (44) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: Content-Length: 0 (17) [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: (0) [Jan 10 17:56:56] --- (9 headers 0 lines) --- [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:2061 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #148 [Jan 10 17:56:56] DEBUG[14576]: chan_sip.c:2071 __sip_ack: Stopping retransmission on '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' of Request 103: Match Not Found [Jan 10 17:56:56] Really destroying SIP dialog '6fd78ff64582f95c3deec38607fdfd19@84.234.24.35' Method: INVITE [Jan 10 17:56:56] Really destroying SIP dialog '7e69b30d05593e6a1482ff6a404075e5@217.14.132.185' Method: BYE [Jan 10 17:57:01] <--- SIP read from 89.213.46.56:64435 ---> NOTIFY sip:84.234.24.35 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-159f327 From: "James Brindle" ;tag=16273a752963b43co3 To: Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1824 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 0: NOTIFY sip:84.234.24.35 SIP/2.0 (31) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-159f327 (58) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 2: From: "James Brindle" ;tag=16273a752963b43co3 (68) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 3: To: (22) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 4: Call-ID: 448cad21-a7370180@192.168.1.211 (40) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 5: CSeq: 1824 NOTIFY (17) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 6: Max-Forwards: 70 (16) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 7: Event: keep-alive (17) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 8: User-Agent: Linksys/SPA941-4.1.15 (33) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 9: Content-Length: 0 (17) [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:4527 parse_request: Header 10: (0) [Jan 10 17:57:01] --- (10 headers 0 lines) --- [Jan 10 17:57:01] <--- Transmitting (no NAT) to 89.213.46.56:64435 ---> SIP/2.0 489 Bad event Via: SIP/2.0/UDP 192.168.1.211:5063;branch=z9hG4bK-159f327;received=89.213.46.56 From: "James Brindle" ;tag=16273a752963b43co3 To: ;tag=as3532ce1c Call-ID: 448cad21-a7370180@192.168.1.211 CSeq: 1824 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 10 17:57:01] DEBUG[14576]: chan_sip.c:14631 sipsock_read: Invalid SIP message - rejected , no callid, len 341 *CLI> show hints -= Registered Asterisk Dial Plan Hints =- 1202@internal_all : SIP/1202 State:Unavailable Watchers 0 1201@internal_all : SIP/1201 State:Idle Watchers 0 1200@internal_all : SIP/1200 State:Idle Watchers 0 1100@internal_all : SIP/1100 State:Idle Watchers 0 1003@internal_all : SIP/1003 State:Idle Watchers 0 1002@internal_all : SIP/1002 State:Unavailable Watchers 0 1001@internal_all : SIP/1001 State:InUse Watchers 0 1000@internal_all : SIP/1000 State:Idle Watchers 0 ---------------- - 8 hints registered The 'show hints' command is deprecated and will be removed in a future release. Please use 'core show hints' instead. *CLI> stop now [Jan 10 17:57:10] Beginning asterisk shutdown.... [Jan 10 17:57:10] Executing last minute cleanups [Jan 10 17:57:10] == Destroying musiconhold processes [Jan 10 17:57:10] Asterisk cleanly ending (0). [Jan 10 17:57:10] DEBUG[14565]: asterisk.c:1193 quit_handler: Asterisk ending (0).