[Jan 12 17:27:47] Asterisk SVN--r430470 built by root @ asterisk-2015 on a x86_64 running Linux on 2015-01-12 03:57:12 UTC [Jan 12 17:28:26] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:27] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:29] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:30] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:30] DEBUG[3065] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3065] chan_iax2.c: ip callno count incremented to 18 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3065] chan_iax2.c: Registration created on call 10095 [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: ip callno count incremented to 19 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: Registration created on call 2881 [Jan 12 17:28:30] DEBUG[3060] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: ip callno count incremented to 20 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: Registration created on call 410 [Jan 12 17:28:30] DEBUG[3061] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: ip callno count incremented to 21 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Registration created on call 7472 [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3061] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: ip callno count incremented to 22 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Registration created on call 15243 [Jan 12 17:28:30] DEBUG[3057] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3057] chan_iax2.c: ip callno count incremented to 23 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3057] chan_iax2.c: Registration created on call 12188 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3057] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3060] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3060] chan_iax2.c: ip callno count incremented to 24 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3060] chan_iax2.c: Registration created on call 4522 [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: ip callno count incremented to 25 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: Registration created on call 4386 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3063] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: ip callno count incremented to 26 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3062] chan_iax2.c: Registration created on call 6625 [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: ip callno count incremented to 27 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3058] chan_iax2.c: Registration created on call 15795 [Jan 12 17:28:30] DEBUG[3059] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3059] chan_iax2.c: ip callno count incremented to 28 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3059] chan_iax2.c: Registration created on call 9783 [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: ip callno count incremented to 29 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3056] chan_iax2.c: Registration created on call 3198 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Allocate call number [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: ip callno count incremented to 30 for 27.111.14.68 [Jan 12 17:28:30] DEBUG[3064] chan_iax2.c: Registration created on call 4859 [Jan 12 17:28:31] DEBUG[3060] chan_iax2.c: Allocate call number [Jan 12 17:28:31] DEBUG[3060] chan_iax2.c: ip callno count incremented to 31 for 27.111.14.68 [Jan 12 17:28:31] DEBUG[3060] chan_iax2.c: Registration created on call 11244 [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: Allocate call number [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: ip callno count incremented to 32 for 27.111.14.68 [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: Registration created on call 8565 [Jan 12 17:28:31] DEBUG[3062] chan_iax2.c: Allocate call number [Jan 12 17:28:31] DEBUG[3062] chan_iax2.c: ip callno count incremented to 33 for 27.111.14.68 [Jan 12 17:28:31] DEBUG[3062] chan_iax2.c: Registration created on call 6868 [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: Allocate call number [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: ip callno count incremented to 34 for 27.111.14.68 [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: Registration created on call 8349 [Jan 12 17:28:31] DEBUG[3065] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3056] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3064] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3061] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3063] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3062] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3058] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3065] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:28:31] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:33] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:34] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:35] Asterisk SVN--r430470 built by root @ asterisk-2015 on a x86_64 running Linux on 2015-01-12 03:57:12 UTC [Jan 12 17:28:35] DEBUG[4059] config.c: Parsing /etc/asterisk/logger.conf [Jan 12 17:28:36] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:37] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:38] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:39] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:28:39] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:28:39] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:28:40] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 33 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 32 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 31 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 30 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 29 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 28 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 27 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 26 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 25 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 24 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 23 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 22 for 27.111.14.68 [Jan 12 17:28:40] DEBUG[3054] chan_iax2.c: ip callno count decremented to 21 for 27.111.14.68 [Jan 12 17:28:41] DEBUG[3054] chan_iax2.c: ip callno count decremented to 20 for 27.111.14.68 [Jan 12 17:28:41] DEBUG[3054] chan_iax2.c: ip callno count decremented to 19 for 27.111.14.68 [Jan 12 17:28:41] DEBUG[3054] chan_iax2.c: ip callno count decremented to 18 for 27.111.14.68 [Jan 12 17:28:41] DEBUG[3054] chan_iax2.c: ip callno count decremented to 17 for 27.111.14.68 [Jan 12 17:28:41] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:43] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:44] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:46] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:47] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:48] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:50] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:51] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 34 instead [Jan 12 17:28:53] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:54] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:28:54] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.183:5060 ---> NOTIFY sip:192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-6c8cd3aa From: 701 ;tag=ade37f6cb3c3dadao1 To: Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 140998 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 <-------------> [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 0 [ 31]: NOTIFY sip:192.168.5.47 SIP/2.0 [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-6c8cd3aa [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 2 [ 55]: From: 701 ;tag=ade37f6cb3c3dadao1 [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 3 [ 22]: To: [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 4 [ 40]: Call-ID: fa228644-5ae8c1e2@192.168.5.183 [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 5 [ 19]: CSeq: 140998 NOTIFY [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 8 [ 36]: User-Agent: Linksys/PAP2T-3.1.15(LS) [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:28:54] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Got NOTIFY Event: keep-alive [Jan 12 17:28:54] VERBOSE[3052] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-6c8cd3aa;received=192.168.5.183 From: 701 ;tag=ade37f6cb3c3dadao1 To: ;tag=as577431f3 Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 140998 NOTIFY Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:28:54] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:28:54] VERBOSE[3052] chan_sip.c: Scheduling destruction of SIP dialog 'fa228644-5ae8c1e2@192.168.5.183' in 32000 ms (Method: NOTIFY) [Jan 12 17:28:55] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:28:57] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 37 instead [Jan 12 17:28:58] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:00] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:01] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:03] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:04] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:05] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:07] DEBUG[3058] chan_iax2.c: ip callno count incremented to 18 for 27.111.14.68 [Jan 12 17:29:07] DEBUG[3021] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Function CALLERID(number) result is '021624717' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Set' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Goto' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'HINT' is 'SIP/707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Gosub' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] app_stack.c: Channel IAX2/099129107-10551 has no datastore, so we're allocating one. [Jan 12 17:29:07] DEBUG[4108][C-0000000b] app_stack.c: Setting 'ARG1' to 'SIP/707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'NoOp' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Answer' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_iax2.c: Answering IAX2 call [Jan 12 17:29:07] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for IAX2 - 099129107 [Jan 12 17:29:07] DEBUG[3033] chan_iax2.c: Checking device state for device 099129107 [Jan 12 17:29:07] DEBUG[3033] chan_iax2.c: Found peer. What's device state of 099129107? addr=27.111.14.68:4569, defaddr=(null) maxms=0, lastms=0 [Jan 12 17:29:07] DEBUG[3033] devicestate.c: Changing state for IAX2/099129107 - state 2 (In use) [Jan 12 17:29:07] DEBUG[3087] app_queue.c: Device 'IAX2/099129107' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jan 12 17:29:07] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:07] DEBUG[3061] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:07] DEBUG[3061] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:07] DEBUG[3061] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:07] DEBUG[3063] chan_iax2.c: Ooh, voice format changed to 'g722' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'EXTEN' is '707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Set' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'ARG1' is 'SIP/707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Set' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'ARG2' is NULL [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Set' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'ext' is '707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'cntx' is '' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Function ISNULL() result is '1' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Expression result is '0' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'cntx' is '' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Function IF(0?@) result is '' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Set' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Result of 'dev' is 'SIP/707' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] pbx.c: Launching 'Dial' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Asked to create a SIP channel with formats: (g722) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Allocating new SIP dialog for 1fd80c322d8ec1482db261214310bb5c@127.0.1.1:5060 - INVITE (No RTP) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] res_rtp_asterisk.c: Allocated port 19218 for RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] rtp_engine.c: RTP instance '0x7ffa3000d9e8' is setup and ready to go [Jan 12 17:29:07] DEBUG[4108][C-0000000b] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Setting NAT on RTP to Off [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 12 17:29:07] DEBUG[4108][C-0000000b] acl.c: For destination '192.168.5.107', our source address is '192.168.5.47'. [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.5.47:5060 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Setting NAT on RTP to Off [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: SIP call-id changed from '1fd80c322d8ec1482db261214310bb5c@127.0.1.1:5060' to '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** Our native formats are (g722) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** Joint capabilities are (g722) [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** Our capabilities are (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** AST_CODEC_CHOOSE formats are g722 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** Our preferred formats from the incoming channel are (g722) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: This channel will not be able to handle video. [Jan 12 17:29:07] DEBUG[4108][C-0000000b] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] channel_internal_api.c: Channel Call ID changing from [C-0000000b] to [C-0000000b] [Jan 12 17:29:07] DEBUG[4108][C-0000000b] rtp_engine.c: Can't find native functions for channel 'IAX2/099129107-10551' [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Outgoing Call for 707 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: This call needs video offers, but there's no video support enabled! [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: We think we can do text [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: This call needs text offers, but there's no text support enabled ! [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: ** Our capability: (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) Video flag: False Text flag: False [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: ** Our prefcodec: (g722) [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Audio is at 19218 [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Done building SDP. Settling with this capability: (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Initializing initreq for method INVITE - callid 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 0 [ 41]: INVITE sip:707@192.168.5.107:5060 SIP/2.0 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 3 [ 62]: From: "6421624717" ;tag=as68de609c [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 4 [ 32]: To: [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 5 [ 42]: Contact: [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 6 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 8 [ 37]: User-Agent: Asterisk PBX SVN--r430470 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 9 [ 35]: Date: Mon, 12 Jan 2015 04:29:07 GMT [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 12 17:29:07] VERBOSE[4108][C-0000000b] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.107:5060: INVITE sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 Max-Forwards: 70 From: "6421624717" ;tag=as68de609c To: Contact: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r430470 Date: Mon, 12 Jan 2015 04:29:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1097 v=0 o=root 479982219 479982219 IN IP4 192.168.5.47 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.47 t=0 0 m=audio 19218 RTP/AVP 9 4 0 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #395 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:07] DEBUG[4108][C-0000000b] chan_iax2.c: Callno 10551: Config blocked sending control frame 22. [Jan 12 17:29:07] DEBUG[4108][C-0000000b] channel.c: IAX2/099129107-10551: Dropping redundant connected line update "Chris Wiltshire" <707>. [Jan 12 17:29:07] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 100 Trying To: From: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 1 [ 32]: To: [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 2 [ 62]: From: "6421624717" ;tag=as68de609c [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 6 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 12 17:29:07] VERBOSE[3052] chan_sip.c: --- (8 headers 0 lines) --- [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking To) --From tag as68de609c --To-tag [Jan 12 17:29:07] DEBUG[3052][C-0000000b] chan_sip.c: *** SIP TIMER: Cancelling retransmission #395 - INVITE (got response) [Jan 12 17:29:07] DEBUG[3052][C-0000000b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' Request 102: Found [Jan 12 17:29:07] DEBUG[3052][C-0000000b] chan_sip.c: SIP response 100 to standard invite [Jan 12 17:29:07] DEBUG[3062] chan_iax2.c: Created trunk peer for '27.111.14.68:4569' [Jan 12 17:29:07] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 180 Ringing To: ;tag=641d5f7eb383343di0 From: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 Contact: "Anonymous" Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=641d5f7eb383343di0 [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 2 [ 62]: From: "6421624717" ;tag=as68de609c [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 6 [ 49]: Contact: "Anonymous" [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 12 17:29:07] VERBOSE[3052] chan_sip.c: --- (9 headers 0 lines) --- [Jan 12 17:29:07] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking To) --From tag as68de609c --To-tag 641d5f7eb383343di0 [Jan 12 17:29:07] DEBUG[3052][C-0000000b] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' Request 102: Found [Jan 12 17:29:07] DEBUG[3052][C-0000000b] chan_sip.c: SIP response 180 to standard invite [Jan 12 17:29:07] VERBOSE[3052][C-0000000b] sip/route.c: sip_route_dump: route/path hop: [Jan 12 17:29:07] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:07] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:07] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:07] DEBUG[3064] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:07] DEBUG[3064] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:08] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 200 OK To: ;tag=641d5f7eb383343di0 From: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 Contact: "Anonymous" Server: Linksys/SPA942-6.1.3(a) Content-Length: 212 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 27200292 27200292 IN IP4 192.168.5.107 s=- c=IN IP4 192.168.5.107 t=0 0 m=audio 16386 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=641d5f7eb383343di0 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 2 [ 62]: From: "6421624717" ;tag=as68de609c [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK6e6b9b71 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 6 [ 49]: Contact: "Anonymous" [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 8 [ 19]: Content-Length: 212 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 10 [ 19]: Supported: replaces [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 12 [ 0]: [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 1 [ 42]: o=- 27200292 27200292 IN IP4 192.168.5.107 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.107 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 5 [ 27]: m=audio 16386 RTP/AVP 0 101 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 9 [ 10]: a=ptime:30 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: --- (12 headers 11 lines) --- [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking To) --From tag as68de609c --To-tag 641d5f7eb383343di0 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Acked pending invite 102 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Stopping retransmission on '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' of Request 102: Match Found [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: SIP response 200 to standard invite [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP o=- 27200292 27200292 IN IP4 192.168.5.107... OK. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107' into... [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port ''. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.107... OK. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] rtp_engine.c: Setting payload 0 (0x7ffa287c5358) based on m type on 0x7ffa002784b0 [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] rtp_engine.c: Setting payload 101 (0x7ffa288545d8) based on m type on 0x7ffa002784b0 [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:09] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Peer audio RTP is at port 192.168.5.107:16386 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] rtp_engine.c: Copying payload 0 (0x7ffa288653e8) from 0x7ffa002784b0 to 0x7ffa3000dbb0 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] rtp_engine.c: Copying payload 101 (0x7ffa287c5358) from 0x7ffa002784b0 to 0x7ffa3000dbb0 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (g722) [Jan 12 17:29:09] DEBUG[3052][C-0000000b] channel.c: Set channel SIP/707-0000000e to read format g722 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] channel.c: Set channel SIP/707-0000000e to write format g722 [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] sip/route.c: sip_route_dump: route/path hop: [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Strict routing enforced for session 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:09] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: set_destination: set destination to 192.168.5.107:5060 [Jan 12 17:29:09] VERBOSE[3052][C-0000000b] chan_sip.c: Transmitting (no NAT) to 192.168.5.107:5060: ACK sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK393893bb Max-Forwards: 70 From: "6421624717" ;tag=as68de609c To: ;tag=641d5f7eb383343di0 Contact: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r430470 Content-Length: 0 --- [Jan 12 17:29:09] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'ACK sip:707' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:09] DEBUG[4108][C-0000000b] channel.c: IAX2/099129107-10551: Dropping redundant connected line update "Chris Wiltshire" <707>. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] features.c: Removing dialed interfaces datastore on SIP/707-0000000e since we're bridging [Jan 12 17:29:09] DEBUG[3021] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 12 17:29:09] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:09] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:09] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge_native_rtp.c: Bridge '60ed6b8b-4654-4421-971c-b402040688fe' can not use native RTP bridge as two channels are required [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: calling simple_bridge technology constructor [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: calling simple_bridge technology start [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30015bf8(IAX2/099129107-10551) is joining [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: pushing 0x7ffa30015bf8(IAX2/099129107-10551) [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge_native_rtp.c: Bridge '60ed6b8b-4654-4421-971c-b402040688fe' can not use native RTP bridge as two channels are required [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is already using the new technology. [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is happy that channel IAX2/099129107-10551 already has read format g722 [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is happy that channel IAX2/099129107-10551 already has write format g722 [Jan 12 17:29:09] DEBUG[4108][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30015bf8(IAX2/099129107-10551) is joining simple_bridge technology [Jan 12 17:29:09] DEBUG[4108][C-0000000b] chan_iax2.c: Callno 10551: Blocked sending control frame 26. [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30001b18(SIP/707-0000000e) is joining [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: pushing 0x7ffa30001b18(SIP/707-0000000e) [Jan 12 17:29:09] DEBUG[3035] cdr.c: Finalized CDR for SIP/707-0000000e - start 1421036947.540000 answer 1421036949.116507 end 1421036949.121527 dispo ANSWERED [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge_native_rtp.c: Bridge '60ed6b8b-4654-4421-971c-b402040688fe' can not use native RTP bridge as it was forbidden while getting details [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is already using the new technology. [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is happy that channel SIP/707-0000000e already has read format g722 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is happy that channel SIP/707-0000000e already has write format g722 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30001b18(SIP/707-0000000e) is joining simple_bridge technology [Jan 12 17:29:09] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 12 17:29:09] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Jan 12 17:29:09] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:09] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: 0x7ffa3000e340 -- Probation learning mode pass with source address 192.168.5.107:16386 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4108][C-0000000b] chan_iax2.c: Expanded trunk '27.111.14.68:4569' to 6400 bytes [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.183:5060 ---> NOTIFY sip:192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-9b36c808 From: 701 ;tag=ade37f6cb3c3dadao1 To: Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 140999 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 <-------------> [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 0 [ 31]: NOTIFY sip:192.168.5.47 SIP/2.0 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-9b36c808 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 2 [ 55]: From: 701 ;tag=ade37f6cb3c3dadao1 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 3 [ 22]: To: [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 4 [ 40]: Call-ID: fa228644-5ae8c1e2@192.168.5.183 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 5 [ 19]: CSeq: 140999 NOTIFY [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 8 [ 36]: User-Agent: Linksys/PAP2T-3.1.15(LS) [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Got NOTIFY Event: keep-alive [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-9b36c808;received=192.168.5.183 From: 701 ;tag=ade37f6cb3c3dadao1 To: ;tag=as577431f3 Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 140999 NOTIFY Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:29:09] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:29:09] VERBOSE[3052] chan_sip.c: Scheduling destruction of SIP dialog 'fa228644-5ae8c1e2@192.168.5.183' in 32000 ms (Method: NOTIFY) [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:09] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:10] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:11] DEBUG[4112][C-0000000b] translate.c: Sample size different 240 vs 480 [Jan 12 17:29:11] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> INVITE sip:021624717@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-ccf81029 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Anonymous" Expires: 30 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 230 Content-Type: application/sdp v=0 o=- 27200292 27200293 IN IP4 192.168.5.107 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 16386 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 0 [ 46]: INVITE sip:021624717@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-ccf81029 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 2 [ 52]: From: ;tag=641d5f7eb383343di0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 3 [ 60]: To: "6421624717" ;tag=as68de609c [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 4 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 5 [ 16]: CSeq: 101 INVITE [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 7 [ 49]: Contact: "Anonymous" [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 8 [ 11]: Expires: 30 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 10 [ 19]: Content-Length: 230 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 12 [ 0]: [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 1 [ 42]: o=- 27200292 27200293 IN IP4 192.168.5.107 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 3 [ 16]: c=IN IP4 0.0.0.0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 5 [ 29]: m=audio 16386 RTP/AVP 0 8 101 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=ptime:30 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Body 11 [ 10]: a=sendonly [Jan 12 17:29:11] VERBOSE[3052] chan_sip.c: --- (12 headers 12 lines) --- [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking From) --From tag 641d5f7eb383343di0 --To-tag as68de609c [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 12 17:29:11] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:11] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Sending to 192.168.5.107:5060 (no NAT) [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Initializing initreq for method INVITE - callid 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP o=- 27200292 27200293 IN IP4 192.168.5.107... OK. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] netsock2.c: Splitting '0.0.0.0' into... [Jan 12 17:29:11] DEBUG[3052][C-0000000b] netsock2.c: ...host '0.0.0.0' and port ''. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP c=IN IP4 0.0.0.0... OK. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Setting payload 0 (0x7ffa287b0d78) based on m type on 0x7ffa00279000 [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found RTP audio format 8 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Setting payload 8 (0x7ffa287abe38) based on m type on 0x7ffa00279000 [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Setting payload 101 (0x7ffa287ccb48) based on m type on 0x7ffa00279000 [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:11] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Peer audio RTP is at port 0.0.0.0:16386 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Copying payload 0 (0x7ffa287c55f8) from 0x7ffa00279000 to 0x7ffa3000dbb0 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Copying payload 8 (0x7ffa287b0d78) from 0x7ffa00279000 to 0x7ffa3000dbb0 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] rtp_engine.c: Copying payload 101 (0x7ffa287abe38) from 0x7ffa00279000 to 0x7ffa3000dbb0 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Jan 12 17:29:11] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Got a SIP re-invite for call 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: SIP/707-0000000e: This call is UP.... [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-ccf81029;received=192.168.5.107 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 101 INVITE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Setting framing from config on incoming call [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: ** Our prefcodec: (g722) [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Audio is at 19218 [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Jan 12 17:29:11] VERBOSE[3052][C-0000000b] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-ccf81029;received=192.168.5.107 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 101 INVITE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 1097 v=0 o=root 479982219 479982220 IN IP4 192.168.5.47 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.47 t=0 0 m=audio 19218 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=recvonly <------------> [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #399 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4108][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to write format slin [Jan 12 17:29:11] DEBUG[4108][C-0000000b] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 12 17:29:11] DEBUG[4108][C-0000000b] chan_iax2.c: Callno 10551: Blocked sending control frame 20. [Jan 12 17:29:11] DEBUG[4108][C-0000000b] chan_iax2.c: Callno 10551: Blocked sending control frame 32. [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> ACK sip:021624717@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-bc86c517 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 101 ACK Max-Forwards: 70 Contact: "Anonymous" User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 0 [ 43]: ACK sip:021624717@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-bc86c517 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 2 [ 52]: From: ;tag=641d5f7eb383343di0 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 3 [ 60]: To: "6421624717" ;tag=as68de609c [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 4 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 5 [ 13]: CSeq: 101 ACK [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 7 [ 49]: Contact: "Anonymous" [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:29:11] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:29:11] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking From) --From tag 641d5f7eb383343di0 --To-tag as68de609c [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #399 [Jan 12 17:29:11] DEBUG[3052][C-0000000b] chan_sip.c: Stopping retransmission on '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' of Response 101: Match Found [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:11] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:12] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> INVITE sip:706@192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-f7e3a894 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: Call-ID: 584f460c-98b8d174@localhost CSeq: 101 INVITE Max-Forwards: 70 Contact: "Anonymous" Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 350 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 27200913 27200913 IN IP4 192.168.5.107 s=- c=IN IP4 192.168.5.107 t=0 0 m=audio 16388 RTP/AVP 0 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 35]: INVITE sip:706@192.168.5.47 SIP/2.0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-f7e3a894 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 63]: From: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 26]: To: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 16]: CSeq: 101 INVITE [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [ 49]: Contact: "Anonymous" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 12]: Expires: 240 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 10 [ 19]: Content-Length: 350 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 11 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 12 [ 19]: Supported: replaces [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 14 [ 0]: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 1 [ 42]: o=- 27200913 27200913 IN IP4 192.168.5.107 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.107 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 5 [ 41]: m=audio 16388 RTP/AVP 0 8 18 96 97 98 101 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 8 [ 22]: a=rtpmap:18 G729a/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 9 [ 24]: a=rtpmap:96 G726-40/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 10 [ 24]: a=rtpmap:97 G726-24/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 11 [ 24]: a=rtpmap:98 G726-16/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 14 [ 10]: a=ptime:30 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 15 [ 10]: a=sendrecv [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (14 headers 16 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking From) --From tag 549a7d2ca40204d4o0 --To-tag [Jan 12 17:29:13] DEBUG[3052] acl.c: For destination '192.168.5.107', our source address is '192.168.5.47'. [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[3052] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:13] DEBUG[3052] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: Sending to 192.168.5.107:5060 (no NAT) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Allocating new SIP dialog for 584f460c-98b8d174@localhost - INVITE (No RTP) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 12 17:29:13] DEBUG[3052][C-0000000c] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces" [Jan 12 17:29:13] DEBUG[3052][C-0000000c] sip/reqresp_parser.c: Found SIP option: -replaces- [Jan 12 17:29:13] DEBUG[3052][C-0000000c] sip/reqresp_parser.c: Matched SIP option: replaces [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Sending to 192.168.5.107:5060 (no NAT) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Initializing initreq for method INVITE - callid 584f460c-98b8d174@localhost [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Using INVITE request as basis request - 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found peer '707' for '707' from 192.168.5.107:5060 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-f7e3a894;received=192.168.5.107 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2558a992 Call-ID: 584f460c-98b8d174@localhost CSeq: 101 INVITE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5470c65e" Content-Length: 0 <------------> [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #400 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Scheduling destruction of SIP dialog '584f460c-98b8d174@localhost' in 32000 ms (Method: INVITE) [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> ACK sip:706@192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-f7e3a894 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2558a992 Call-ID: 584f460c-98b8d174@localhost CSeq: 101 ACK Max-Forwards: 70 Contact: "Anonymous" User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 32]: ACK sip:706@192.168.5.47 SIP/2.0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-f7e3a894 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 63]: From: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 41]: To: ;tag=as2558a992 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 13]: CSeq: 101 ACK [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [ 49]: Contact: "Anonymous" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking From) --From tag 549a7d2ca40204d4o0 --To-tag as2558a992 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #400 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '584f460c-98b8d174@localhost' of Response 101: Match Found [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> INVITE sip:706@192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4fe338dc From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: Call-ID: 584f460c-98b8d174@localhost CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47",algorithm=MD5,response="79931790cab812050310f2f75f92d95e" Contact: "Anonymous" Expires: 240 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 350 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 27200913 27200913 IN IP4 192.168.5.107 s=- c=IN IP4 192.168.5.107 t=0 0 m=audio 16388 RTP/AVP 0 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 35]: INVITE sip:706@192.168.5.47 SIP/2.0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4fe338dc [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 63]: From: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 26]: To: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [155]: Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47",algorithm=MD5,response="79931790cab812050310f2f75f92d95e" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 49]: Contact: "Anonymous" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 9 [ 12]: Expires: 240 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 10 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 11 [ 19]: Content-Length: 350 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 12 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 13 [ 19]: Supported: replaces [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 15 [ 0]: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 1 [ 42]: o=- 27200913 27200913 IN IP4 192.168.5.107 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.107 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 5 [ 41]: m=audio 16388 RTP/AVP 0 8 18 96 97 98 101 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 8 [ 22]: a=rtpmap:18 G729a/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 9 [ 24]: a=rtpmap:96 G726-40/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 10 [ 24]: a=rtpmap:97 G726-24/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 11 [ 24]: a=rtpmap:98 G726-16/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 14 [ 10]: a=ptime:30 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 15 [ 10]: a=sendrecv [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (15 headers 16 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking From) --From tag 549a7d2ca40204d4o0 --To-tag [Jan 12 17:29:13] DEBUG[3052] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] DEBUG[3052] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Sending to 192.168.5.107:5060 (no NAT) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Initializing initreq for method INVITE - callid 584f460c-98b8d174@localhost [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Using INVITE request as basis request - 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found peer '707' for '707' from 192.168.5.107:5060 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7ffa288dacd8' [Jan 12 17:29:13] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Allocated port 10126 for RTP instance '0x7ffa288dacd8' [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: RTP instance '0x7ffa288dacd8' is setup and ready to go [Jan 12 17:29:13] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa288dacd8' [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Setting NAT on RTP to Off [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP o=- 27200913 27200913 IN IP4 192.168.5.107... OK. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port ''. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.107... OK. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 0 (0x7ffa287c92d8) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 8 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 8 (0x7ffa287c9b18) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 18 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 18 (0x7ffa287ca358) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 96 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 96 (0x7ffa287b37f8) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 97 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 97 (0x7ffa287b4038) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 98 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 98 (0x7ffa287b4878) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 101 (0x7ffa287b50b8) based on m type on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format G729a for ID 18 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Unsetting payload 96 on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found unknown media description format G726-40 for ID 96 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Unsetting payload 97 on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found unknown media description format G726-24 for ID 97 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Unsetting payload 98 on 0x7ffa00279000 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found unknown media description format G726-16 for ID 98 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729) [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa288dacd8' [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Peer audio RTP is at port 192.168.5.107:16388 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa287b5518) from 0x7ffa00279000 to 0x7ffa288daea0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 8 (0x7ffa287c92d8) from 0x7ffa00279000 to 0x7ffa288daea0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 18 (0x7ffa287c9b18) from 0x7ffa00279000 to 0x7ffa288daea0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa287b4878) from 0x7ffa00279000 to 0x7ffa288daea0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ffa288dacd8' [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Checking SIP call limits for device 707 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Updating call counter for incoming call [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.47' into... [Jan 12 17:29:13] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.47' and port ''. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Looking for 706 in numberplan-IAX2-SetDDI (domain 192.168.5.47) [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** Our native formats are (ulaw) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** Joint capabilities are (ulaw) [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** Our capabilities are (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: This channel will not be able to handle video. [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] sip/route.c: sip_route_dump: route/path hop: [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: SIP/707-0000000f: New call is still down.... Trying... [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4fe338dc;received=192.168.5.107 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: Call-ID: 584f460c-98b8d174@localhost CSeq: 102 INVITE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:13] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:13] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:13] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'NoOp' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Not taking any branch [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Not taking any branch [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Not taking any branch [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Not taking any branch [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Not taking any branch [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '1' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'GotoIf' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function CALLERID(num) result is '707' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Set' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'EXTEN' is '706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Goto' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'HINT' is 'SIP/706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Gosub' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] app_stack.c: Channel SIP/707-0000000f has no datastore, so we're allocating one. [Jan 12 17:29:13] DEBUG[4116][C-0000000c] app_stack.c: Setting 'ARG1' to 'SIP/706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'NoOp' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Answer' [Jan 12 17:29:13] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:13] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:13] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: SIP answering channel: SIP/707-0000000f [Jan 12 17:29:13] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Setting framing from config on incoming call [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: ** Our prefcodec: (nothing) [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Audio is at 10126 [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4fe338dc;received=192.168.5.107 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 102 INVITE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 1095 v=0 o=root 27533907 27533907 IN IP4 192.168.5.47 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.47 t=0 0 m=audio 10126 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv <------------> [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #403 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> ACK sip:706@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-7f50fcea From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47",algorithm=MD5,response="79931790cab812050310f2f75f92d95e" Contact: "Anonymous" User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 37]: ACK sip:706@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-7f50fcea [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 63]: From: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 41]: To: ;tag=as2d9cca6b [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 13]: CSeq: 102 ACK [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [155]: Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47",algorithm=MD5,response="79931790cab812050310f2f75f92d95e" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 49]: Contact: "Anonymous" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 9 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (11 headers 0 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking From) --From tag 549a7d2ca40204d4o0 --To-tag as2d9cca6b [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #403 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '584f460c-98b8d174@localhost' of Response 102: Match Found [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: 0x7ffa287c56a0 -- Probation learning mode pass with source address 192.168.5.107:16388 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'EXTEN' is '706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Set' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'ARG1' is 'SIP/706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Set' [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'ARG2' is NULL [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Set' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'ext' is '706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'cntx' is '' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function ISNULL() result is '1' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Expression result is '0' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'cntx' is '' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Function IF(0?@) result is '' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Set' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Result of 'dev' is 'SIP/706' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] pbx.c: Launching 'Dial' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Asked to create a SIP channel with formats: (ulaw) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Allocating new SIP dialog for 781dba536d7d76545d41826c3e5cd3ce@127.0.1.1:5060 - INVITE (No RTP) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7ffa300263e8' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Allocated port 15030 for RTP instance '0x7ffa300263e8' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: RTP instance '0x7ffa300263e8' is setup and ready to go [Jan 12 17:29:13] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa300263e8' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Setting NAT on RTP to Off [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 12 17:29:13] DEBUG[4116][C-0000000c] acl.c: For destination '192.168.5.106', our source address is '192.168.5.47'. [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Setting AST_TRANSPORT_UDP with address 192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Setting NAT on RTP to Off [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: SIP call-id changed from '781dba536d7d76545d41826c3e5cd3ce@127.0.1.1:5060' to '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** Our native formats are (ulaw) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** Joint capabilities are (ulaw) [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** Our capabilities are (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** Our preferred formats from the incoming channel are (ulaw) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: This channel will not be able to handle video. [Jan 12 17:29:13] DEBUG[4116][C-0000000c] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] channel_internal_api.c: Channel Call ID changing from [C-0000000c] to [C-0000000c] [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa287b5518) from 0x7ffa288daea0 to 0x7ffa300265b0 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Copying payload 8 (0x7ffa287c92d8) from 0x7ffa288daea0 to 0x7ffa300265b0 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Copying payload 18 (0x7ffa287c9b18) from 0x7ffa288daea0 to 0x7ffa300265b0 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa287b4878) from 0x7ffa288daea0 to 0x7ffa300265b0 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Seeded SDP of 'SIP/706-00000010' with that of 'SIP/707-0000000f' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Outgoing Call for 706 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: This call needs video offers, but there's no video support enabled! [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: We think we can do text [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: This call needs text offers, but there's no text support enabled ! [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: ** Our capability: (ulaw|g722|g723|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) Video flag: False Text flag: False [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: ** Our prefcodec: (ulaw) [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Audio is at 15030 [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Done building SDP. Settling with this capability: (ulaw|g722|g723|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Initializing initreq for method INVITE - callid 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 0 [ 41]: INVITE sip:706@192.168.5.106:5060 SIP/2.0 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 3 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 4 [ 32]: To: [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 5 [ 36]: Contact: [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 6 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 8 [ 37]: User-Agent: Asterisk PBX SVN--r430470 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 9 [ 35]: Date: Mon, 12 Jan 2015 04:29:13 GMT [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 12 17:29:13] VERBOSE[4116][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.106:5060: INVITE sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r430470 Date: Mon, 12 Jan 2015 04:29:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1099 v=0 o=root 1584042168 1584042168 IN IP4 192.168.5.47 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.47 t=0 0 m=audio 15030 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #405 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:13] DEBUG[4116][C-0000000c] channel.c: SIP/707-0000000f: Dropping redundant connected line update "Avijit Khetarpal" <706>. [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 100 Trying To: From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 32]: To: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (8 headers 0 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: *** SIP TIMER: Cancelling retransmission #405 - INVITE (got response) [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' Request 102: Found [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 100 to standard invite [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 180 Ringing To: ;tag=f58370b417a2215di0 From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 Contact: "Avi" Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 43]: Contact: "Avi" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (9 headers 0 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag f58370b417a2215di0 [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' Request 102: Found [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 180 to standard invite [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] sip/route.c: sip_route_dump: route/path hop: [Jan 12 17:29:13] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 706 [Jan 12 17:29:13] DEBUG[3033] chan_sip.c: Checking device state for peer 706 [Jan 12 17:29:13] DEBUG[3033] devicestate.c: Changing state for SIP/706 - state 1 (Not in use) [Jan 12 17:29:13] DEBUG[4116][C-0000000c] rtp_engine.c: Setting early bridge SDP of 'SIP/707-0000000f' with that of 'SIP/706-00000010' [Jan 12 17:29:13] DEBUG[4116][C-0000000c] channel.c: Driver for channel 'SIP/707-0000000f' does not support indication 3, emulating it [Jan 12 17:29:13] DEBUG[4116][C-0000000c] channel.c: Set channel SIP/707-0000000f to write format slin [Jan 12 17:29:13] DEBUG[4116][C-0000000c] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> INFO sip:706@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-c4f91f6e From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 103 INFO Max-Forwards: 70 Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47:5060",algorithm=MD5,response="58a5c02e2ce8c6750f10f756c3dfd07b" User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 24 Content-Type: application/dtmf-relay Signal=# Duration=100 <-------------> [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 0 [ 38]: INFO sip:706@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-c4f91f6e [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 2 [ 63]: From: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 3 [ 41]: To: ;tag=as2d9cca6b [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 4 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 5 [ 14]: CSeq: 103 INFO [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 7 [160]: Authorization: Digest username="707",realm="asterisk",nonce="5470c65e",uri="sip:706@192.168.5.47:5060",algorithm=MD5,response="58a5c02e2ce8c6750f10f756c3dfd07b" [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 8 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 9 [ 18]: Content-Length: 24 [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 10 [ 36]: Content-Type: application/dtmf-relay [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Header 11 [ 0]: [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 0 [ 8]: Signal=# [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: Body 1 [ 12]: Duration=100 [Jan 12 17:29:13] VERBOSE[3052] chan_sip.c: --- (11 headers 2 lines) --- [Jan 12 17:29:13] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking From) --From tag 549a7d2ca40204d4o0 --To-tag as2d9cca6b [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: Receiving INFO! [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: * DTMF-relay event received: # [Jan 12 17:29:13] VERBOSE[3052][C-0000000c] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-c4f91f6e;received=192.168.5.107 From: "Anonymous" ;tag=549a7d2ca40204d4o0 To: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 103 INFO Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:29:13] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:13] DTMF[4116][C-0000000c] channel.c: DTMF end '#' received on SIP/707-0000000f, duration 100 ms [Jan 12 17:29:13] DTMF[4116][C-0000000c] channel.c: DTMF begin emulation of '#' with duration 100 queued on SIP/707-0000000f [Jan 12 17:29:13] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DTMF[4116][C-0000000c] channel.c: DTMF end emulation of '#' queued on SIP/707-0000000f [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:14] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 200 OK To: ;tag=f58370b417a2215di0 From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 Contact: "Avi" Server: Linksys/SPA942-6.1.5(a) Content-Length: 214 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 211994626 211994626 IN IP4 192.168.5.106 s=- c=IN IP4 192.168.5.106 t=0 0 m=audio 16404 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK61f33918 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 6 [ 43]: Contact: "Avi" [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 8 [ 19]: Content-Length: 214 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 9 [ 61]: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 10 [ 19]: Supported: replaces [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 12 [ 0]: [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 1 [ 44]: o=- 211994626 211994626 IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 5 [ 27]: m=audio 16404 RTP/AVP 0 101 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 9 [ 10]: a=ptime:30 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: --- (12 headers 11 lines) --- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Acked pending invite 102 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' of Request 102: Match Found [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 200 to standard invite [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP o=- 211994626 211994626 IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port ''. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 0 (0x7ffa287ca738) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 101 (0x7ffa2884f2b8) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa300263e8' [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Peer audio RTP is at port 192.168.5.106:16404 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa2884f718) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa287ca738) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7ffa300263e8' [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] sip/route.c: sip_route_dump: route/path hop: [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Transmitting (no NAT) to 192.168.5.106:5060: ACK sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK121f26a8 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r430470 Content-Length: 0 --- [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'ACK sip:706' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4116][C-0000000c] channel.c: SIP/707-0000000f: Dropping redundant connected line update "Avijit Khetarpal" <706>. [Jan 12 17:29:15] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 706 [Jan 12 17:29:15] DEBUG[3033] chan_sip.c: Checking device state for peer 706 [Jan 12 17:29:15] DEBUG[3033] devicestate.c: Changing state for SIP/706 - state 1 (Not in use) [Jan 12 17:29:15] DEBUG[4116][C-0000000c] rtp_engine.c: Setting early bridge SDP of 'SIP/707-0000000f' with that of 'SIP/706-00000010' [Jan 12 17:29:15] DEBUG[4116][C-0000000c] channel.c: Set channel SIP/707-0000000f to write format ulaw [Jan 12 17:29:15] DEBUG[4116][C-0000000c] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 12 17:29:15] DEBUG[4116][C-0000000c] features.c: Removing dialed interfaces datastore on SIP/706-00000010 since we're bridging [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge_native_rtp.c: Bridge '737861ca-884f-4848-9b1f-e2a4b11d2aea' can not use native RTP bridge as two channels are required [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology softmix has less preference than simple_bridge (10 <= 50). Skipping. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology constructor [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology start [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c768(SIP/707-0000000f) is joining [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pushing 0x7ffa3002c768(SIP/707-0000000f) [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge_native_rtp.c: Bridge '737861ca-884f-4848-9b1f-e2a4b11d2aea' can not use native RTP bridge as two channels are required [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is already using the new technology. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/707-0000000f already has read format ulaw [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/707-0000000f already has write format ulaw [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c768(SIP/707-0000000f) is joining simple_bridge technology [Jan 12 17:29:15] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Changing ssrc from 1889347366 to 908280824 due to a source change [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c5f8(SIP/706-00000010) is joining [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pushing 0x7ffa3002c5f8(SIP/706-00000010) [Jan 12 17:29:15] DEBUG[3035] cdr.c: Finalized CDR for SIP/706-00000010 - start 1421036953.918006 answer 1421036955.527978 end 1421036955.532372 dispo ANSWERED [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Chose bridge technology native_rtp [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling native_rtp technology constructor [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: moving 0x7ffa3002c768(SIP/707-0000000f) to dummy bridge temporarily [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c768(SIP/707-0000000f) is leaving simple_bridge technology (dummy) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology stop [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/706-00000010 already has read format ulaw [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/706-00000010 already has write format ulaw [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c5f8(SIP/706-00000010) is joining native_rtp technology [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Sending reinvite on SIP '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' - It's audio soon redirected to IP 192.168.5.107:16388 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our prefcodec: (ulaw) [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Audio is at 15030 [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Initializing already initialized SIP dialog 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (presumably reinvite) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 0 [ 41]: INVITE sip:706@192.168.5.106:5060 SIP/2.0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK539b2cf6 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 3 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 4 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 5 [ 36]: Contact: [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 6 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 7 [ 16]: CSeq: 103 INVITE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 8 [ 37]: User-Agent: Asterisk PBX SVN--r430470 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.106:5060: INVITE sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK539b2cf6 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 1101 v=0 o=root 1584042168 1584042169 IN IP4 192.168.5.107 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.107 t=0 0 m=audio 16388 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #407 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Sending reinvite on SIP '584f460c-98b8d174@localhost' - It's audio soon redirected to IP 192.168.5.106:16404 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Strict routing enforced for session 584f460c-98b8d174@localhost [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.107:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our prefcodec: (nothing) [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Audio is at 10126 [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Initializing already initialized SIP dialog 584f460c-98b8d174@localhost (presumably reinvite) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 0 [ 41]: INVITE sip:707@192.168.5.107:5060 SIP/2.0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK79c0c581 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 3 [ 43]: From: ;tag=as2d9cca6b [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 4 [ 61]: To: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 5 [ 36]: Contact: [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 6 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 8 [ 37]: User-Agent: Asterisk PBX SVN--r430470 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.107:5060: INVITE sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK79c0c581 Max-Forwards: 70 From: ;tag=as2d9cca6b To: "Anonymous" ;tag=549a7d2ca40204d4o0 Contact: Call-ID: 584f460c-98b8d174@localhost CSeq: 102 INVITE User-Agent: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 1097 v=0 o=root 27533907 27533908 IN IP4 192.168.5.106 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.106 t=0 0 m=audio 16404 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #408 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/707-0000000f already has read format ulaw [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/707-0000000f already has write format ulaw [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c768(SIP/707-0000000f) is joining native_rtp technology [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling native_rtp technology start [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology destructor [Jan 12 17:29:15] DEBUG[4118][C-0000000c] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Chose bridge technology native_rtp [Jan 12 17:29:15] DEBUG[4116][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is already using the new technology. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Chose bridge technology native_rtp [Jan 12 17:29:15] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is already using the new technology. [Jan 12 17:29:15] DEBUG[4118][C-0000000c] res_rtp_asterisk.c: Ooh, format changed from none to ulaw [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 200 OK To: ;tag=f58370b417a2215di0 From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK539b2cf6 Contact: "Avi" Server: Linksys/SPA942-6.1.5(a) Content-Length: 214 Content-Type: application/sdp v=0 o=- 211994626 211994627 IN IP4 192.168.5.106 s=- c=IN IP4 192.168.5.106 t=0 0 m=audio 16404 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK539b2cf6 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 6 [ 43]: Contact: "Avi" [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 8 [ 19]: Content-Length: 214 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 10 [ 0]: [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 1 [ 44]: o=- 211994626 211994627 IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 5 [ 27]: m=audio 16404 RTP/AVP 0 101 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 9 [ 10]: a=ptime:30 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: --- (10 headers 11 lines) --- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Acked pending invite 103 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #407 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' of Request 103: Match Found [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 200 to RE-invite on outgoing call 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP o=- 211994626 211994627 IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port ''. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 0 (0x7ffa287b3c08) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 101 (0x7ffa28856618) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Peer audio RTP is at port 192.168.5.106:16404 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa28856a78) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa287b3c08) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa300263e8' [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Transmitting (no NAT) to 192.168.5.106:5060: ACK sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK71dfcb09 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN--r430470 Content-Length: 0 --- [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'ACK sip:706' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 200 OK To: "Anonymous" ;tag=549a7d2ca40204d4o0 From: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK79c0c581 Contact: "Anonymous" Server: Linksys/SPA942-6.1.3(a) Content-Length: 212 Content-Type: application/sdp v=0 o=- 27200913 27200914 IN IP4 192.168.5.107 s=- c=IN IP4 192.168.5.107 t=0 0 m=audio 16388 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 1 [ 61]: To: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 2 [ 43]: From: ;tag=as2d9cca6b [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 3 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 102 INVITE [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK79c0c581 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 6 [ 49]: Contact: "Anonymous" [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 8 [ 19]: Content-Length: 212 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 10 [ 0]: [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 1 [ 42]: o=- 27200913 27200914 IN IP4 192.168.5.107 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.107 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 5 [ 27]: m=audio 16388 RTP/AVP 0 101 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 9 [ 10]: a=ptime:30 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: --- (10 headers 11 lines) --- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking To) --From tag as2d9cca6b --To-tag 549a7d2ca40204d4o0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Acked pending invite 102 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #408 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '584f460c-98b8d174@localhost' of Request 102: Match Found [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 200 to RE-invite on outgoing call 584f460c-98b8d174@localhost [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP o=- 27200913 27200914 IN IP4 192.168.5.107... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port ''. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.107... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 0 (0x7ffa28856618) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 101 (0x7ffa287ca738) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Peer audio RTP is at port 192.168.5.107:16388 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa287c32a8) from 0x7ffa002784b0 to 0x7ffa288daea0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa28856618) from 0x7ffa002784b0 to 0x7ffa288daea0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa288dacd8' [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Updating call counter for incoming call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Strict routing enforced for session 584f460c-98b8d174@localhost [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.107:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Transmitting (no NAT) to 192.168.5.107:5060: ACK sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK25d3e513 Max-Forwards: 70 From: ;tag=as2d9cca6b To: "Anonymous" ;tag=549a7d2ca40204d4o0 Contact: Call-ID: 584f460c-98b8d174@localhost CSeq: 102 ACK User-Agent: Asterisk PBX SVN--r430470 Content-Length: 0 --- [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'ACK sip:707' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Sending reinvite on SIP '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' - It's audio soon redirected to IP 192.168.5.107:16388 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[4118][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our native-bridge filtered capablity: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: ** Our prefcodec: (ulaw) [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Audio is at 15030 [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ulaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g722 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g723 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec alaw to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec gsm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g726aal2 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec adpcm to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec slin to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec lpc10 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g729 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec speex to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec ilbc to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren7 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec siren14 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec testlaw to SDP [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec g719 to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec opus to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding codec none to SDP [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: -- Done with adding codecs to SDP [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Initializing already initialized SIP dialog 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (presumably reinvite) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 0 [ 41]: INVITE sip:706@192.168.5.106:5060 SIP/2.0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 1 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK342f16a7 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 3 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 4 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 5 [ 36]: Contact: [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 6 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 7 [ 16]: CSeq: 104 INVITE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 8 [ 37]: User-Agent: Asterisk PBX SVN--r430470 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 9 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 10 [ 26]: Supported: replaces, timer [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge) [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] VERBOSE[4118][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.106:5060: INVITE sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK342f16a7 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 1101 v=0 o=root 1584042168 1584042170 IN IP4 192.168.5.107 s=Asterisk PBX SVN--r430470 c=IN IP4 192.168.5.107 t=0 0 m=audio 16388 RTP/AVP 0 9 4 8 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 102 115 116 107 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:122 L16/12000 a=rtpmap:118 L16/16000 a=rtpmap:123 L16/24000 a=rtpmap:124 L16/32000 a=rtpmap:125 L16/44000 a=rtpmap:126 L16/48000 a=rtpmap:127 L16/96000 a=rtpmap:96 L16/192000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 speex/16000 a=rtpmap:119 speex/32000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=0 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:116 G719/48000 a=fmtp:116 bitrate=64000 a=rtpmap:107 opus/48000/2 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:20 a=sendrecv --- [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #409 [Jan 12 17:29:15] DEBUG[4118][C-0000000c] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 200 OK To: ;tag=f58370b417a2215di0 From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 104 INVITE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK342f16a7 Contact: "Avi" Server: Linksys/SPA942-6.1.5(a) Content-Length: 214 Content-Type: application/sdp v=0 o=- 211994626 211994628 IN IP4 192.168.5.106 s=- c=IN IP4 192.168.5.106 t=0 0 m=audio 16404 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 104 INVITE [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK342f16a7 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 6 [ 43]: Contact: "Avi" [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 7 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 8 [ 19]: Content-Length: 214 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Header 10 [ 0]: [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 0 [ 3]: v=0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 1 [ 44]: o=- 211994626 211994628 IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 2 [ 3]: s=- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 3 [ 22]: c=IN IP4 192.168.5.106 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 5 [ 27]: m=audio 16404 RTP/AVP 0 101 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 9 [ 10]: a=ptime:30 [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 12 17:29:15] VERBOSE[3052] chan_sip.c: --- (10 headers 11 lines) --- [Jan 12 17:29:15] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag f58370b417a2215di0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Acked pending invite 104 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #409 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' of Request 104: Match Found [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: SIP response 200 to RE-invite on outgoing call 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP o=- 211994626 211994628 IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port ''. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.106... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 0 (0x7ffa28866a28) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found RTP audio format 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Setting payload 101 (0x7ffa287ca358) based on m type on 0x7ffa002784b0 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] strings.c: failed to extend from 64 to 98 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Capabilities: us - (g722|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Peer audio RTP is at port 192.168.5.106:16404 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 0 (0x7ffa287c9218) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] rtp_engine.c: Copying payload 101 (0x7ffa28866a28) from 0x7ffa002784b0 to 0x7ffa300265b0 [Jan 12 17:29:15] DEBUG[3052][C-0000000c] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7ffa300263e8' [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We're settling with these formats: (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: We have an owner, now see if we need to change this call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Setting native formats after processing SDP. peer joint formats (ulaw), old nativeformats (ulaw) [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:15] DEBUG[3052][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:15] VERBOSE[3052][C-0000000c] chan_sip.c: Transmitting (no NAT) to 192.168.5.106:5060: ACK sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK639ba6c4 Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Contact: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 104 ACK User-Agent: Asterisk PBX SVN--r430470 Content-Length: 0 --- [Jan 12 17:29:15] DEBUG[3052][C-0000000c] chan_sip.c: Trying to put 'ACK sip:706' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:15] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:16] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:17] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:18] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:19] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: ip callno count incremented to 19 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: Registration created on call 4906 [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: ip callno count incremented to 20 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: Registration created on call 11375 [Jan 12 17:29:20] DEBUG[3063] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[3065] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3065] chan_iax2.c: ip callno count incremented to 21 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3065] chan_iax2.c: Registration created on call 15698 [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[3058] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: ip callno count incremented to 22 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3059] chan_iax2.c: Registration created on call 15766 [Jan 12 17:29:20] DEBUG[3064] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[3058] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: No remote address on RTP instance '0x7ffa3000d9e8' so dropping frame [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> REFER sip:021624717@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-47a0fa37 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Referred-By: "Anonymous" Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 REFER Max-Forwards: 70 Contact: "Anonymous" Refer-To: User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 0 [ 45]: REFER sip:021624717@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-47a0fa37 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 2 [ 52]: From: ;tag=641d5f7eb383343di0 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 3 [ 60]: To: "6421624717" ;tag=as68de609c [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 4 [ 47]: Referred-By: "Anonymous" [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 5 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 6 [ 15]: CSeq: 102 REFER [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 8 [ 49]: Contact: "Anonymous" [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 9 [125]: Refer-To: [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 10 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: --- (12 headers 0 lines) --- [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking From) --From tag 641d5f7eb383343di0 --To-tag as68de609c [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: **** Received REFER (9) - Command in SIP REFER [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: Call 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 got a SIP call transfer from caller: (REFER)! [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 584f460c-98b8d174@localhost F-tag: 549a7d2ca40204d4o0 T-tag: as2d9cca6b [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: SIP transfer to extension 706@numberplan-IAX2-SetDDI by 707@192.168.5.47 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: SIP attended transfer: Transferer channel SIP/707-0000000e [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-47a0fa37;received=192.168.5.107 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 102 REFER Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'SIP/2.0 202' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Looking for callid 584f460c-98b8d174@localhost (fromtag 549a7d2ca40204d4o0 totag as2d9cca6b) [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Matched INCOMING call - their tag is 549a7d2ca40204d4o0 Our tag is as2d9cca6b [Jan 12 17:29:20] DEBUG[4118][C-0000000c] res_rtp_asterisk.c: Setting the marker bit due to a source update [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Moving 0x7ffa30015bf8(IAX2/099129107-10551) into bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea swapping with SIP/707-0000000f [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: pulling 0x7ffa30015bf8(IAX2/099129107-10551) [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30015bf8(IAX2/099129107-10551) is leaving simple_bridge technology [Jan 12 17:29:20] DEBUG[3035] cdr.c: Finalized CDR for IAX2/099129107-10551 - start 1421036947.238353 answer 1421036947.243418 end 1421036960.842258 dispo ANSWERED [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pushing 0x7ffa30015bf8(IAX2/099129107-10551) by swapping with 0x7ffa3002c768(SIP/707-0000000f) [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Setting 0x7ffa3002c768(SIP/707-0000000f) state from:0 to:2 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pulling 0x7ffa3002c768(SIP/707-0000000f) [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c768(SIP/707-0000000f) is leaving native_rtp technology [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_native_rtp.c: Discontinued RTP bridging of 'SIP/706-00000010' and 'IAX2/099129107-10551' - media will flow through Asterisk core [Jan 12 17:29:20] DEBUG[3035] cdr.c: Finalized CDR for SIP/707-0000000f - start 1421036953.860229 answer 1421036953.871707 end 1421036960.843758 dispo ANSWERED [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_native_rtp.c: Bridge '737861ca-884f-4848-9b1f-e2a4b11d2aea' can not use native RTP bridge as it was forbidden while getting details [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology constructor [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: moving 0x7ffa3002c5f8(SIP/706-00000010) to dummy bridge temporarily [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c5f8(SIP/706-00000010) is leaving native_rtp technology (dummy) [Jan 12 17:29:20] DEBUG[4108][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to write format g722 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling native_rtp technology stop [Jan 12 17:29:20] DEBUG[4108][C-0000000b] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel IAX2/099129107-10551 already has read format g722 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel IAX2/099129107-10551 already has write format g722 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa30015bf8(IAX2/099129107-10551) is joining simple_bridge technology [Jan 12 17:29:20] DEBUG[3052][C-0000000b] channel.c: Set channel SIP/706-00000010 to read format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to write format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to read format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] channel.c: Set channel SIP/706-00000010 to write format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/706-00000010 already has read format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is happy that channel SIP/706-00000010 already has write format slin16 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c5f8(SIP/706-00000010) is joining simple_bridge technology [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology start [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling native_rtp technology destructor [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_native_rtp.c: Bridge '60ed6b8b-4654-4421-971c-b402040688fe' can not use native RTP bridge as two channels are required [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is already using the new technology. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge_channel.c: Setting 0x7ffa30001b18(SIP/707-0000000e) state from:0 to:2 [Jan 12 17:29:20] DEBUG[4112][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: pulling 0x7ffa30001b18(SIP/707-0000000e) [Jan 12 17:29:20] DEBUG[4112][C-0000000b] bridge_channel.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: 0x7ffa30001b18(SIP/707-0000000e) is leaving simple_bridge technology [Jan 12 17:29:20] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: dissolving bridge with cause 16(Normal Clearing) [Jan 12 17:29:20] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: queueing action type:13 sub:1001 [Jan 12 17:29:20] DEBUG[4112][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe is dissolved, not performing smart bridge operation. [Jan 12 17:29:20] DEBUG[4112][C-0000000b] res_rtp_asterisk.c: Changing ssrc from 675121774 to 987905042 due to a source change [Jan 12 17:29:20] DEBUG[4112][C-0000000b] channel.c: Hanging up channel 'SIP/707-0000000e' [Jan 12 17:29:20] DEBUG[4112][C-0000000b] chan_sip.c: update_call_counter(707) - decrement call limit counter on hangup [Jan 12 17:29:20] DEBUG[4112][C-0000000b] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:20] DEBUG[4112][C-0000000b] chan_sip.c: Call to peer '707' removed from call limit 0 [Jan 12 17:29:20] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:20] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:20] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:20] DEBUG[4112][C-0000000b] chan_sip.c: SIP Transfer: Not hanging up right now... Rescheduling hangup for 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060. [Jan 12 17:29:20] VERBOSE[4112][C-0000000b] chan_sip.c: Scheduling destruction of SIP dialog '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' in 32000 ms (Method: REFER) [Jan 12 17:29:20] DEBUG[4116][C-0000000c] res_rtp_asterisk.c: Changing ssrc from 908280824 to 610441083 due to a source change [Jan 12 17:29:20] DEBUG[4116][C-0000000c] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 12 17:29:20] DEBUG[4116][C-0000000c] pbx.c: Spawn extension (numberplan-cost-efficient,706,50006) exited non-zero on 'SIP/707-0000000f' [Jan 12 17:29:20] DEBUG[4116][C-0000000c] channel.c: Soft-Hanging (0x10) up channel 'SIP/707-0000000f' [Jan 12 17:29:20] DEBUG[4116][C-0000000c] channel.c: Hanging up channel 'SIP/707-0000000f' [Jan 12 17:29:20] DEBUG[4116][C-0000000c] chan_sip.c: Hangup call SIP/707-0000000f, SIP callid 584f460c-98b8d174@localhost [Jan 12 17:29:20] VERBOSE[4116][C-0000000c] chan_sip.c: Scheduling destruction of SIP dialog '584f460c-98b8d174@localhost' in 32000 ms (Method: INFO) [Jan 12 17:29:20] DEBUG[4116][C-0000000c] chan_sip.c: Strict routing enforced for session 584f460c-98b8d174@localhost [Jan 12 17:29:20] VERBOSE[4116][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:20] DEBUG[4116][C-0000000c] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:20] DEBUG[4116][C-0000000c] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:20] VERBOSE[4116][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.107:5060 [Jan 12 17:29:20] VERBOSE[4116][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.107:5060: BYE sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK7370bd80 Max-Forwards: 70 From: ;tag=as2d9cca6b To: "Anonymous" ;tag=549a7d2ca40204d4o0 Call-ID: 584f460c-98b8d174@localhost CSeq: 103 BYE User-Agent: Asterisk PBX SVN--r430470 Proxy-Authorization: Digest username="707", realm="asterisk", algorithm=MD5, uri="sip:192.168.5.47", nonce="5470c65e", response="18b953afe4b2a3d25c472070d4021078" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 12 17:29:20] DEBUG[4116][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #412 [Jan 12 17:29:20] DEBUG[4116][C-0000000c] chan_sip.c: Trying to put 'BYE sip:707' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge_native_rtp.c: Bridge '737861ca-884f-4848-9b1f-e2a4b11d2aea' can not use native RTP bridge as it was forbidden while getting details [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge.c: Bridge technology softmix does not have any capabilities we want. [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge.c: Chose bridge technology simple_bridge [Jan 12 17:29:20] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is already using the new technology. [Jan 12 17:29:20] DEBUG[3021] threadpool.c: Increasing threadpool stasis-core's size by 1 [Jan 12 17:29:20] DEBUG[4108][C-0000000b] chan_iax2.c: Callno 10551: Config blocked sending control frame 22. [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: actually destroying basic bridge, nobody wants it anymore [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: calling basic bridge destructor [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: calling simple_bridge technology stop [Jan 12 17:29:20] DEBUG[3052][C-0000000b] bridge.c: Bridge 60ed6b8b-4654-4421-971c-b402040688fe: calling simple_bridge technology destructor [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Strict routing enforced for session 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:20] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:20] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: set_destination: set destination to 192.168.5.107:5060 [Jan 12 17:29:20] VERBOSE[3052][C-0000000b] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.107:5060: NOTIFY sip:707@192.168.5.107:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK44bc17d8 Max-Forwards: 70 From: "6421624717" ;tag=as68de609c To: ;tag=641d5f7eb383343di0 Contact: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 103 NOTIFY User-Agent: Asterisk PBX SVN--r430470 Event: refer;id=102 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 16 SIP/2.0 200 OK --- [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #413 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:20] DEBUG[3035] cdr.c: Finalized CDR for IAX2/099129107-10551 - start 1421036960.843047 answer 1421036960.843047 end 1421036960.853482 dispo ANSWERED [Jan 12 17:29:20] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:20] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:20] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:20] DEBUG[3035] cdr.c: CDR for SIP/707-0000000e is dialed and has no Party B; discarding [Jan 12 17:29:20] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:20] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:20] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 200 OK To: "Anonymous" ;tag=549a7d2ca40204d4o0 From: ;tag=as2d9cca6b Call-ID: 584f460c-98b8d174@localhost CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK7370bd80 Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 1 [ 61]: To: "Anonymous" ;tag=549a7d2ca40204d4o0 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 2 [ 43]: From: ;tag=as2d9cca6b [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 3 [ 36]: Call-ID: 584f460c-98b8d174@localhost [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 4 [ 13]: CSeq: 103 BYE [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK7370bd80 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 6 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: --- (8 headers 0 lines) --- [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: = Looking for Call ID: 584f460c-98b8d174@localhost (Checking To) --From tag as2d9cca6b --To-tag 549a7d2ca40204d4o0 [Jan 12 17:29:20] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #412 [Jan 12 17:29:20] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '584f460c-98b8d174@localhost' of Request 103: Match Found [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Destroying SIP dialog 584f460c-98b8d174@localhost [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: Really destroying SIP dialog '584f460c-98b8d174@localhost' Method: INFO [Jan 12 17:29:20] DEBUG[3052] rtp_engine.c: Destroyed RTP instance '0x7ffa288dacd8' [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> SIP/2.0 200 OK To: ;tag=641d5f7eb383343di0 From: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 103 NOTIFY Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK44bc17d8 Server: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 1 [ 50]: To: ;tag=641d5f7eb383343di0 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 2 [ 62]: From: "6421624717" ;tag=as68de609c [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 4 [ 16]: CSeq: 103 NOTIFY [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK44bc17d8 [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 6 [ 31]: Server: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 12 17:29:20] VERBOSE[3052] chan_sip.c: --- (8 headers 0 lines) --- [Jan 12 17:29:20] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking To) --From tag as68de609c --To-tag 641d5f7eb383343di0 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #413 [Jan 12 17:29:20] DEBUG[3052][C-0000000b] chan_sip.c: Stopping retransmission on '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' of Request 103: Match Found [Jan 12 17:29:20] DEBUG[4118][C-0000000c] res_rtp_asterisk.c: Difference is 42512, ms is 5334 [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: ip callno count incremented to 23 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: Registration created on call 6671 [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: ip callno count incremented to 24 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: Registration created on call 215 [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[3060] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:20] DEBUG[3058] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3058] chan_iax2.c: ip callno count incremented to 25 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3058] chan_iax2.c: Registration created on call 11912 [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: Allocate call number [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: ip callno count incremented to 26 for 27.111.14.68 [Jan 12 17:29:20] DEBUG[3061] chan_iax2.c: Registration created on call 11337 [Jan 12 17:29:20] DEBUG[3057] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3056] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3063] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3063] chan_iax2.c: ip callno count incremented to 27 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3063] chan_iax2.c: Registration created on call 5709 [Jan 12 17:29:21] DEBUG[3062] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3062] chan_iax2.c: ip callno count incremented to 28 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3062] chan_iax2.c: Registration created on call 5885 [Jan 12 17:29:21] DEBUG[3057] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3057] chan_iax2.c: ip callno count incremented to 29 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3057] chan_iax2.c: Registration created on call 7506 [Jan 12 17:29:21] DEBUG[3059] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3059] chan_iax2.c: ip callno count incremented to 30 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3059] chan_iax2.c: Registration created on call 5550 [Jan 12 17:29:21] DEBUG[3065] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3065] chan_iax2.c: ip callno count incremented to 31 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3065] chan_iax2.c: Registration created on call 9958 [Jan 12 17:29:21] DEBUG[3060] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3060] chan_iax2.c: ip callno count incremented to 32 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3060] chan_iax2.c: Registration created on call 2452 [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: ip callno count incremented to 33 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: Registration created on call 13472 [Jan 12 17:29:21] DEBUG[3064] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3064] chan_iax2.c: ip callno count incremented to 34 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3064] chan_iax2.c: Registration created on call 14564 [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: Allocate call number [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: ip callno count incremented to 35 for 27.111.14.68 [Jan 12 17:29:21] DEBUG[3058] chan_iax2.c: Registration created on call 13124 [Jan 12 17:29:21] DEBUG[3056] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3056] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3063] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3062] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3061] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3057] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3059] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3065] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3064] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:21] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:22] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:24] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.107:5060 ---> BYE sip:021624717@192.168.5.47:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4e453794 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 103 BYE Max-Forwards: 70 User-Agent: Linksys/SPA942-6.1.3(a) Content-Length: 0 <-------------> [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 0 [ 43]: BYE sip:021624717@192.168.5.47:5060 SIP/2.0 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4e453794 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 2 [ 52]: From: ;tag=641d5f7eb383343di0 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 3 [ 60]: To: "6421624717" ;tag=as68de609c [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 4 [ 59]: Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 7 [ 35]: User-Agent: Linksys/SPA942-6.1.3(a) [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: --- (9 headers 0 lines) --- [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: = Looking for Call ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 (Checking From) --From tag 641d5f7eb383343di0 --To-tag as68de609c [Jan 12 17:29:24] DEBUG[3052][C-0000000b] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 12 17:29:24] DEBUG[3052][C-0000000b] chan_sip.c: Initializing initreq for method BYE - callid 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:24] DEBUG[3052][C-0000000b] netsock2.c: Splitting '192.168.5.107:5060' into... [Jan 12 17:29:24] DEBUG[3052][C-0000000b] netsock2.c: ...host '192.168.5.107' and port '5060'. [Jan 12 17:29:24] VERBOSE[3052][C-0000000b] chan_sip.c: Sending to 192.168.5.107:5060 (no NAT) [Jan 12 17:29:24] DEBUG[3052][C-0000000b] chan_sip.c: Setting SIP_ALREADYGONE on dialog 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:24] DEBUG[3052][C-0000000b] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:24] VERBOSE[3052][C-0000000b] chan_sip.c: Scheduling destruction of SIP dialog '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' in 32000 ms (Method: BYE) [Jan 12 17:29:24] DEBUG[3052][C-0000000b] chan_sip.c: Received bye, no owner, selfdestruct soon. [Jan 12 17:29:24] VERBOSE[3052][C-0000000b] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.107:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.107:5060;branch=z9hG4bK-4e453794;received=192.168.5.107 From: ;tag=641d5f7eb383343di0 To: "6421624717" ;tag=as68de609c Call-ID: 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 CSeq: 103 BYE Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:29:24] DEBUG[3052][C-0000000b] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.107:5060 [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.183:5060 ---> NOTIFY sip:192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-7a77351f From: 701 ;tag=ade37f6cb3c3dadao1 To: Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 141000 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 <-------------> [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 0 [ 31]: NOTIFY sip:192.168.5.47 SIP/2.0 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 1 [ 59]: Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-7a77351f [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 2 [ 55]: From: 701 ;tag=ade37f6cb3c3dadao1 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 3 [ 22]: To: [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 4 [ 40]: Call-ID: fa228644-5ae8c1e2@192.168.5.183 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 5 [ 19]: CSeq: 141000 NOTIFY [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 8 [ 36]: User-Agent: Linksys/PAP2T-3.1.15(LS) [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Got NOTIFY Event: keep-alive [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-7a77351f;received=192.168.5.183 From: 701 ;tag=ade37f6cb3c3dadao1 To: ;tag=as577431f3 Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 141000 NOTIFY Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:29:24] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:29:24] VERBOSE[3052] chan_sip.c: Scheduling destruction of SIP dialog 'fa228644-5ae8c1e2@192.168.5.183' in 32000 ms (Method: NOTIFY) [Jan 12 17:29:25] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:26] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:28] DEBUG[3064] chan_iax2.c: JB STATS:IAX2/099129107-10551 ping=8 ljitterms=-1 ljbdelayms=0 ltotlost=-1 lrecentlosspct=-1 ldropped=0 looo=-1 lrecvd=-1 rjitterms=0 rjbdelayms=40 rtotlost=0 rrecentlosspct=0 rdropped=0 rooo=0 rrecvd=1 [Jan 12 17:29:28] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:29] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 34 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 33 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 32 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 31 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 30 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 29 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 28 for 27.111.14.68 [Jan 12 17:29:30] DEBUG[3054] chan_iax2.c: ip callno count decremented to 27 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 26 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 25 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 24 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 23 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 22 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 21 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 20 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 19 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3054] chan_iax2.c: ip callno count decremented to 18 for 27.111.14.68 [Jan 12 17:29:31] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:31] DEBUG[3066] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 4 instead [Jan 12 17:29:32] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:33] DEBUG[3059] chan_iax2.c: Callno 10551: Blocked receiving control frame 20. [Jan 12 17:29:33] DEBUG[3065] chan_iax2.c: Immediately destroying 10551, having received hangup [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Setting 0x7ffa30015bf8(IAX2/099129107-10551) state from:0 to:1 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pulling 0x7ffa30015bf8(IAX2/099129107-10551) [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa30015bf8(IAX2/099129107-10551) is leaving simple_bridge technology [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: dissolving bridge with cause 16(Normal Clearing) [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Setting 0x7ffa3002c5f8(SIP/706-00000010) state from:0 to:2 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: queueing action type:13 sub:1001 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is dissolved, not performing smart bridge operation. [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge is returning 0x7ffa30015bf8(IAX2/099129107-10551) to read format g722 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to read format g722 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] bridge_channel.c: Bridge is returning 0x7ffa30015bf8(IAX2/099129107-10551) to write format g722 [Jan 12 17:29:33] DEBUG[4108][C-0000000b] channel.c: Set channel IAX2/099129107-10551 to write format g722 [Jan 12 17:29:33] DEBUG[3035] cdr.c: Finalized CDR for IAX2/099129107-10551 - start 1421036960.843035 answer 1421036960.843035 end 1421036973.694526 dispo ANSWERED [Jan 12 17:29:33] DEBUG[4108][C-0000000b] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 12 17:29:33] DEBUG[4108][C-0000000b] pbx.c: Spawn extension (default,707,50006) exited non-zero on 'IAX2/099129107-10551' [Jan 12 17:29:33] DEBUG[4108][C-0000000b] channel.c: Soft-Hanging (0x10) up channel 'IAX2/099129107-10551' [Jan 12 17:29:33] DEBUG[4108][C-0000000b] channel.c: Hanging up channel 'IAX2/099129107-10551' [Jan 12 17:29:33] DEBUG[4108][C-0000000b] chan_iax2.c: We're hanging up IAX2/099129107-10551 now... [Jan 12 17:29:33] DEBUG[4108][C-0000000b] chan_iax2.c: Really destroying IAX2/099129107-10551 now... [Jan 12 17:29:33] DEBUG[4108][C-0000000b] chan_iax2.c: schedule decrement of callno used for 27.111.14.68 in 60 seconds [Jan 12 17:29:33] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for IAX2 - 099129107 [Jan 12 17:29:33] DEBUG[3033] chan_iax2.c: Checking device state for device 099129107 [Jan 12 17:29:33] DEBUG[3033] chan_iax2.c: Found peer. What's device state of 099129107? addr=27.111.14.68:4569, defaddr=(null) maxms=0, lastms=0 [Jan 12 17:29:33] DEBUG[3033] devicestate.c: Changing state for IAX2/099129107 - state 0 (Unknown) [Jan 12 17:29:33] DEBUG[3087] app_queue.c: Device 'IAX2/099129107' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: pulling 0x7ffa3002c5f8(SIP/706-00000010) [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: 0x7ffa3002c5f8(SIP/706-00000010) is leaving simple_bridge technology [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea is dissolved, not performing smart bridge operation. [Jan 12 17:29:33] DEBUG[4118][C-0000000c] res_rtp_asterisk.c: Changing ssrc from 932976392 to 7934367 due to a source change [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge is returning 0x7ffa3002c5f8(SIP/706-00000010) to read format ulaw [Jan 12 17:29:33] DEBUG[4118][C-0000000c] channel.c: Set channel SIP/706-00000010 to read format ulaw [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge_channel.c: Bridge is returning 0x7ffa3002c5f8(SIP/706-00000010) to write format ulaw [Jan 12 17:29:33] DEBUG[4118][C-0000000c] channel.c: Set channel SIP/706-00000010 to write format ulaw [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: actually destroying basic bridge, nobody wants it anymore [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling basic bridge destructor [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology stop [Jan 12 17:29:33] DEBUG[4118][C-0000000c] bridge.c: Bridge 737861ca-884f-4848-9b1f-e2a4b11d2aea: calling simple_bridge technology destructor [Jan 12 17:29:33] DEBUG[4118][C-0000000c] channel.c: Hanging up channel 'SIP/706-00000010' [Jan 12 17:29:33] DEBUG[4118][C-0000000c] chan_sip.c: Hangup call SIP/706-00000010, SIP callid 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:33] VERBOSE[4118][C-0000000c] chan_sip.c: Scheduling destruction of SIP dialog '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' in 32000 ms (Method: INVITE) [Jan 12 17:29:33] DEBUG[4118][C-0000000c] chan_sip.c: Strict routing enforced for session 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:33] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 12 17:29:33] DEBUG[4118][C-0000000c] netsock2.c: Splitting '192.168.5.106:5060' into... [Jan 12 17:29:33] DEBUG[4118][C-0000000c] netsock2.c: ...host '192.168.5.106' and port '5060'. [Jan 12 17:29:33] VERBOSE[4118][C-0000000c] chan_sip.c: set_destination: set destination to 192.168.5.106:5060 [Jan 12 17:29:33] VERBOSE[4118][C-0000000c] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.5.106:5060: BYE sip:706@192.168.5.106:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK2c2a55bd Max-Forwards: 70 From: "Chris Wiltshire" ;tag=as68d121d5 To: ;tag=f58370b417a2215di0 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 105 BYE User-Agent: Asterisk PBX SVN--r430470 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 12 17:29:33] DEBUG[4118][C-0000000c] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #417 [Jan 12 17:29:33] DEBUG[4118][C-0000000c] chan_sip.c: Trying to put 'BYE sip:706' onto UDP socket destined for 192.168.5.106:5060 [Jan 12 17:29:33] DEBUG[3035] cdr.c: CDR for SIP/706-00000010 is dialed and has no Party B; discarding [Jan 12 17:29:33] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 706 [Jan 12 17:29:33] DEBUG[3033] chan_sip.c: Checking device state for peer 706 [Jan 12 17:29:33] DEBUG[3033] devicestate.c: Changing state for SIP/706 - state 1 (Not in use) [Jan 12 17:29:33] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.106:5060 ---> SIP/2.0 200 OK To: ;tag=f58370b417a2215di0 From: "Chris Wiltshire" ;tag=as68d121d5 Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 CSeq: 105 BYE Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK2c2a55bd Server: Linksys/SPA942-6.1.5(a) Content-Length: 0 <-------------> [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 1 [ 55]: To: ;tag=f58370b417a2215di0 [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 2 [ 61]: From: "Chris Wiltshire" ;tag=as68d121d5 [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 3 [ 59]: Call-ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 4 [ 13]: CSeq: 105 BYE [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 5 [ 57]: Via: SIP/2.0/UDP 192.168.5.47:5060;branch=z9hG4bK2c2a55bd [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 6 [ 31]: Server: Linksys/SPA942-6.1.5(a) [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 12 17:29:33] VERBOSE[3052] chan_sip.c: --- (8 headers 0 lines) --- [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: = Looking for Call ID: 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 (Checking To) --From tag as68d121d5 --To-tag f58370b417a2215di0 [Jan 12 17:29:33] DEBUG[3052][C-0000000c] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #417 [Jan 12 17:29:33] DEBUG[3052][C-0000000c] chan_sip.c: Stopping retransmission on '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' of Request 105: Match Found [Jan 12 17:29:33] DEBUG[3052] chan_sip.c: Destroying SIP dialog 520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060 [Jan 12 17:29:33] VERBOSE[3052] chan_sip.c: Really destroying SIP dialog '520c0e31625c8db418a4b5f767f0a976@192.168.5.47:5060' Method: INVITE [Jan 12 17:29:33] DEBUG[3052] rtp_engine.c: Destroyed RTP instance '0x7ffa300263e8' [Jan 12 17:29:34] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:35] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:36] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:38] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:39] DEBUG[3066] chan_iax2.c: Dropping unused iax2 trunk peer '27.111.14.68:4569' [Jan 12 17:29:39] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:39] VERBOSE[3052] chan_sip.c: <--- SIP read from UDP:192.168.5.183:5060 ---> NOTIFY sip:192.168.5.47 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-85458d1 From: 701 ;tag=ade37f6cb3c3dadao1 To: Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 141001 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 <-------------> [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 0 [ 31]: NOTIFY sip:192.168.5.47 SIP/2.0 [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-85458d1 [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 2 [ 55]: From: 701 ;tag=ade37f6cb3c3dadao1 [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 3 [ 22]: To: [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 4 [ 40]: Call-ID: fa228644-5ae8c1e2@192.168.5.183 [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 5 [ 19]: CSeq: 141001 NOTIFY [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 7 [ 17]: Event: keep-alive [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 8 [ 36]: User-Agent: Linksys/PAP2T-3.1.15(LS) [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 12 17:29:39] VERBOSE[3052] chan_sip.c: --- (10 headers 0 lines) --- [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Got NOTIFY Event: keep-alive [Jan 12 17:29:39] VERBOSE[3052] chan_sip.c: <--- Transmitting (no NAT) to 192.168.5.183:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.183:5060;branch=z9hG4bK-85458d1;received=192.168.5.183 From: 701 ;tag=ade37f6cb3c3dadao1 To: ;tag=as577431f3 Call-ID: fa228644-5ae8c1e2@192.168.5.183 CSeq: 141001 NOTIFY Server: Asterisk PBX SVN--r430470 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Jan 12 17:29:39] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:29:39] VERBOSE[3052] chan_sip.c: Scheduling destruction of SIP dialog 'fa228644-5ae8c1e2@192.168.5.183' in 32000 ms (Method: NOTIFY) [Jan 12 17:29:40] DEBUG[4111] threadpool.c: Worker thread idle timeout reached. Dying. [Jan 12 17:29:40] DEBUG[3021] threadpool.c: Destroying worker thread 15 [Jan 12 17:29:40] DEBUG[4123] threadpool.c: Worker thread idle timeout reached. Dying. [Jan 12 17:29:40] DEBUG[3021] threadpool.c: Destroying worker thread 16 [Jan 12 17:29:41] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:42] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:43] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:45] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 37 instead [Jan 12 17:29:46] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:48] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 37 instead [Jan 12 17:29:49] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:51] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:52] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:29:53] DEBUG[4107] threadpool.c: Worker thread idle timeout reached. Dying. [Jan 12 17:29:53] DEBUG[3021] threadpool.c: Destroying worker thread 14 [Jan 12 17:29:53] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:54] DEBUG[3052] chan_sip.c: = Looking for Call ID: fa228644-5ae8c1e2@192.168.5.183 (Checking From) --From tag ade37f6cb3c3dadao1 --To-tag [Jan 12 17:29:54] DEBUG[3052] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY [Jan 12 17:29:54] DEBUG[3052] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.183:5060 [Jan 12 17:29:55] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:56] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:56] DEBUG[3052] chan_sip.c: Auto destroying SIP dialog '4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060' [Jan 12 17:29:56] DEBUG[3052] chan_sip.c: Destroying SIP dialog 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:56] DEBUG[3052] chan_sip.c: Updating call counter for outgoing call [Jan 12 17:29:56] DEBUG[3052] chan_sip.c: This call did not properly clean up call limits. Call ID 4ce527a87e4754a34d129d6e3907d0c7@192.168.5.47:5060 [Jan 12 17:29:56] DEBUG[3033] devicestate.c: No provider found, checking channel drivers for SIP - 707 [Jan 12 17:29:56] DEBUG[3033] chan_sip.c: Checking device state for peer 707 [Jan 12 17:29:56] DEBUG[3033] devicestate.c: Changing state for SIP/707 - state 1 (Not in use) [Jan 12 17:29:56] DEBUG[3052] rtp_engine.c: Destroyed RTP instance '0x7ffa3000d9e8' [Jan 12 17:29:58] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:29:59] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead [Jan 12 17:30:01] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 35 instead [Jan 12 17:30:02] DEBUG[3048] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 36 instead