[2011-01-14 10:36:32] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Set' [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:32] -- Executing [comp@DID_ngn:1] Set("SIP/ngn54-00005953", "LINE=1234552") in new stack [2011-01-14 10:36:32] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Goto' [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:32] -- Executing [comp@DID_ngn:4] Goto("SIP/ngn54-00005953", "IVR_1234552,s,1") in new stack [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:9420 pbx_builtin_goto: [2011-01-14 10:36:32] -- Goto (IVR_1234552,s,1) [2011-01-14 10:36:32] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Set' [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:32] -- Executing [s@IVR_1234552:1] Set("SIP/ngn54-00005953", "NUMINVALID=0") in new stack [2011-01-14 10:36:32] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Set' [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:32] -- Executing [s@IVR_1234552:2] Set("SIP/ngn54-00005953", "NUMTIMEOUTS=0") in new stack [2011-01-14 10:36:32] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Playback' [2011-01-14 10:36:32] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:32] -- Executing [s@IVR_1234552:3] Playback("SIP/ngn54-00005953", "menu-33") in new stack [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:6020 sip_answer: SIP answering channel: SIP/ngn54-00005953 [2011-01-14 10:36:32] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:10913 transmit_response_with_sdp: Setting framing from config on incoming call [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:10559 add_sdp: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:10560 add_sdp: ** Our prefcodec: 0x0 (nothing) [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:10669 add_sdp: -- Done with adding codecs to SDP [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:10808 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-01-14 10:36:32] DEBUG[18498]: chan_sip.c:3089 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 62.148.237.132:5060 [2011-01-14 10:36:33] DEBUG[18498]: channel.c:2830 __ast_answer: Didn't receive a media frame from SIP/ngn54-00005953 within 500 ms of answering. Continuing anyway [2011-01-14 10:36:33] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format gsm [2011-01-14 10:36:33] DEBUG[18498]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to alaw [2011-01-14 10:36:33] DEBUG[18498]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160 [2011-01-14 10:36:33] DEBUG[18498]: res_rtp_asterisk.c:1140 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x7fca702fe968' [2011-01-14 10:36:33] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:36:33] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:36:33] -- Playing 'menu-33.gsm' (language 'ru') [2011-01-14 10:36:33] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:34] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:35] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:36] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:37] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:38] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:39] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:39] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:39] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:39] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:39] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format alaw [2011-01-14 10:36:39] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Set' [2011-01-14 10:36:39] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:39] -- Executing [s@IVR_1234552:4] Set("SIP/ngn54-00005953", "TIMEOUT(digit)=3") in new stack [2011-01-14 10:36:39] VERBOSE[18498]: func_timeout.c:179 timeout_write: [2011-01-14 10:36:39] -- Digit timeout set to 3.000 [2011-01-14 10:36:39] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Set' [2011-01-14 10:36:39] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:39] -- Executing [s@IVR_1234552:5] Set("SIP/ngn54-00005953", "TIMEOUT(response)=10") in new stack [2011-01-14 10:36:39] VERBOSE[18498]: func_timeout.c:171 timeout_write: [2011-01-14 10:36:39] -- Response timeout set to 10.000 [2011-01-14 10:36:39] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'BackGround' [2011-01-14 10:36:39] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:39] -- Executing [s@IVR_1234552:6] BackGround("SIP/ngn54-00005953", "menu-34") in new stack [2011-01-14 10:36:39] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format gsm [2011-01-14 10:36:39] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:36:39] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:36:39] -- Playing 'menu-34.gsm' (language 'ru') [2011-01-14 10:36:39] DEBUG[18498]: res_timing_timerfd.c:178 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [2011-01-14 10:36:39] DEBUG[18498]: res_timing_timerfd.c:178 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [2011-01-14 10:36:40] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:40] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:41] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:43] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:44] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:44] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:45] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 62.148.237.195:55680 [2011-01-14 10:36:46] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/ngn54-00005953 [2011-01-14 10:36:46] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/ngn54-00005953 [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 62.148.237.195:55680 [2011-01-14 10:36:46] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/ngn54-00005953, duration 330 ms [2011-01-14 10:36:46] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/ngn54-00005953 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:46] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format alaw [2011-01-14 10:36:46] DEBUG[18498]: pbx.c:4067 pbx_extension_helper: Launching 'Queue' [2011-01-14 10:36:46] VERBOSE[18498]: pbx.c:4075 pbx_extension_helper: [2011-01-14 10:36:46] -- Executing [1@IVR_1234552:1] Queue("SIP/ngn54-00005953", "IVR-52-1,trR,,,1800") in new stack [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:5670 queue_exec: NO QUEUE_PRIO variable found. Using default. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:5720 queue_exec: queue: IVR-52-1, options: trR, url: , announce: , expires: 1294985206, priority: 0 [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:2245 update_realtime_members: Queue IVR-52-1 has no realtime members defined. No need for update [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:2356 join_queue: Queue 'IVR-52-1' Join, Channel 'SIP/ngn54-00005953', Position '1' [2011-01-14 10:36:46] DEBUG[18498]: channel.c:4340 ast_indicate_data: Driver for channel 'SIP/ngn54-00005953' does not support indication 3, emulating it [2011-01-14 10:36:46] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format slin [2011-01-14 10:36:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3688 is_our_turn: There is 1 available member. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3702 is_our_turn: It's our turn (SIP/ngn54-00005953). [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:4264 try_calling: SIP/ngn54-00005953 is trying to call a queue member. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3886 calc_metric: Disregarding penalty, 0 members and 0 in penaltymemberslimit. [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d50usoltceva@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d50usoltceva@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d50kozhaeva@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d50kozhaeva@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'SIP/user5' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:24997 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:7207 sip_alloc: Allocating new SIP dialog for 02cc143f61cc65bf16e62fd052d64bc2@91.216.53.76:0 - INVITE (No RTP) [2011-01-14 10:36:46] DEBUG[18498]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fca78387b38' [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 14704 for RTP instance '0x7fca78387b38' [2011-01-14 10:36:46] DEBUG[18498]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x7fca78387b38' is setup and ready to go [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fca78387b38' [2011-01-14 10:36:46] VERBOSE[18498]: netsock2.c:499 ast_set_qos: [2011-01-14 10:36:46] == Using SIP RTP CoS mark 5 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:4675 do_setnat: Setting NAT on RTP to Off [2011-01-14 10:36:46] DEBUG[18498]: acl.c:715 ast_ouraddrfor: For destination '10.1.16.100', our source address is '91.216.53.28'. [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 91.216.53.28:5060 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6549 sip_new: *** Our native formats are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6550 sip_new: *** Joint capabilities are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6551 sip_new: *** Our capabilities are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6552 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6554 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6582 sip_new: This channel will not be able to handle video. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:5204 sip_call: Outgoing Call for user5 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:5446 update_call_counter: Updating call counter for outgoing call [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10559 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10560 add_sdp: ** Our prefcodec: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10669 add_sdp: -- Done with adding codecs to SDP [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10808 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method INVITE - callid 18eaaa191c4911775fd755474a383414@91.216.53.28:5060 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.16.100:56909 [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'SIP/user4' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:24997 sip_request_call: Asked to create a SIP channel with formats: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:7207 sip_alloc: Allocating new SIP dialog for 69ee07600be8832b756ab370043695ff@91.216.53.76:0 - INVITE (No RTP) [2011-01-14 10:36:46] DEBUG[18498]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fca7854f528' [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 19152 for RTP instance '0x7fca7854f528' [2011-01-14 10:36:46] DEBUG[18498]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x7fca7854f528' is setup and ready to go [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fca7854f528' [2011-01-14 10:36:46] VERBOSE[18498]: netsock2.c:499 ast_set_qos: [2011-01-14 10:36:46] == Using SIP RTP CoS mark 5 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:4675 do_setnat: Setting NAT on RTP to Off [2011-01-14 10:36:46] DEBUG[18498]: acl.c:715 ast_ouraddrfor: For destination '10.1.16.97', our source address is '91.216.53.28'. [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 91.216.53.28:5060 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6549 sip_new: *** Our native formats are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6550 sip_new: *** Joint capabilities are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6551 sip_new: *** Our capabilities are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6552 sip_new: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6554 sip_new: *** Our preferred formats from the incoming channel are 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:6582 sip_new: This channel will not be able to handle video. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:5204 sip_call: Outgoing Call for user4 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:5446 update_call_counter: Updating call counter for outgoing call [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10559 add_sdp: ** Our capability: 0x8 (alaw) Video flag: False Text flag: False [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10560 add_sdp: ** Our prefcodec: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10669 add_sdp: -- Done with adding codecs to SDP [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:10808 add_sdp: Done building SDP. Settling with this capability: 0x8 (alaw) [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:2823 initialize_initreq: Initializing initreq for method INVITE - callid 72414b7f092963a84f81480156e67ecd@91.216.53.28:5060 [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:3089 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 10.1.16.97:2414 [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d25user2@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d25user2@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d25user1@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d25user1@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d25user6@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d25user6@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: app_queue.c:3048 ring_one: (Parallel) Trying 'LOCAL/9d25user3@default' with metric 0 [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable BACKGROUNDSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable PLAYBACKSTATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMTIMEOUTS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable NUMINVALID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable AUTO_MONITOR. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable MYSQL_STATUS. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable fetchid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable resultid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable connid. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable LINE. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPCALLID. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [2011-01-14 10:36:46] DEBUG[18498]: channel.c:5845 ast_channel_inherit_variables: Not copying variable SIPURI. [2011-01-14 10:36:46] DEBUG[18498]: chan_local.c:243 local_devicestate: Checking if extension 9d25user3@default exists (devicestate) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:36:46] DEBUG[18498]: channel.c:3492 ast_read_generator_actions: Generator got voice, switching to phase locked mode [2011-01-14 10:36:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:36:46] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:36:46] -- SIP/user4-0000595e is ringing [2011-01-14 10:36:46] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:36:46] -- SIP/user5-0000595d is ringing [2011-01-14 10:36:46] VERBOSE[18498]: app_queue.c:3511 wait_for_answer: [2011-01-14 10:36:46] -- SIP/user5-0000595d is busy [2011-01-14 10:36:46] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'SIP/user5-0000595d' [2011-01-14 10:36:46] DEBUG[18498]: chan_sip.c:5819 sip_hangup: Hangup call SIP/user5-0000595d, SIP callid 18eaaa191c4911775fd755474a383414@91.216.53.28:5060 [2011-01-14 10:36:46] DEBUG[18498]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fca78387b38' [2011-01-14 10:36:46] VERBOSE[18498]: app_queue.c:3193 rna: [2011-01-14 10:36:46] -- Nobody picked up in 0 ms [2011-01-14 10:36:47] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:48] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:49] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:50] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:51] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:52] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:53] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:54] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:55] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:56] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:58] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:36:59] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:00] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:01] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:02] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:02] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:03] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:04] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:05] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:06] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:07] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:07] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:08] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:09] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:11] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3530 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user6@default-4833;1 is circuit-busy [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3193 rna: [2011-01-14 10:37:11] -- Nobody picked up in 25000 ms [2011-01-14 10:37:11] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'Local/9d25user6@default-4833;1' [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user1@default-fd9b;1 is ringing [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user3@default-c1ca;1 is ringing [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user2@default-d548;1 is ringing [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user1@default-fd9b;1 is ringing [2011-01-14 10:37:11] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:11] -- Local/9d25user2@default-d548;1 is ringing [2011-01-14 10:37:11] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:12] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:12] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:13] VERBOSE[18498]: app_queue.c:3548 wait_for_answer: [2011-01-14 10:37:13] -- Local/9d25user3@default-c1ca;1 is ringing [2011-01-14 10:37:13] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:13] DEBUG[18498]: app_queue.c:3598 wait_for_answer: Dunno what to do with control type -1 [2011-01-14 10:37:13] VERBOSE[18498]: app_queue.c:3530 wait_for_answer: [2011-01-14 10:37:13] -- Local/9d25user2@default-d548;1 is circuit-busy [2011-01-14 10:37:13] VERBOSE[18498]: app_queue.c:3193 rna: [2011-01-14 10:37:13] -- Nobody picked up in 27000 ms [2011-01-14 10:37:13] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'Local/9d25user2@default-d548;1' [2011-01-14 10:37:14] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:15] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:16] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:17] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:18] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:19] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:20] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:22] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:23] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:24] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:25] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:26] VERBOSE[18498]: app_queue.c:3564 wait_for_answer: [2011-01-14 10:37:26] -- Local/9d25user3@default-c1ca;1 connected line has changed. Saving it until answer for SIP/ngn54-00005953 [2011-01-14 10:37:26] DEBUG[18498]: app_queue.c:3598 wait_for_answer: Dunno what to do with control type -1 [2011-01-14 10:37:26] VERBOSE[18498]: app_queue.c:3484 wait_for_answer: [2011-01-14 10:37:26] -- Local/9d25user3@default-c1ca;1 answered SIP/ngn54-00005953 [2011-01-14 10:37:26] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'Local/9d50usoltceva@default-25fa;1' [2011-01-14 10:37:26] DEBUG[18498]: chan_local.c:864 local_hangup: This local call has the ANSWERED_ELSEWHERE flag set. [2011-01-14 10:37:26] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'Local/9d50kozhaeva@default-c126;1' [2011-01-14 10:37:26] DEBUG[18498]: chan_local.c:864 local_hangup: This local call has the ANSWERED_ELSEWHERE flag set. [2011-01-14 10:37:26] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'SIP/user4-0000595e' [2011-01-14 10:37:26] DEBUG[18498]: chan_sip.c:5783 sip_hangup: This call was answered elsewhere [2011-01-14 10:37:26] DEBUG[18498]: chan_sip.c:5819 sip_hangup: Hangup call SIP/user4-0000595e, SIP callid 72414b7f092963a84f81480156e67ecd@91.216.53.28:5060 [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fca7854f528' [2011-01-14 10:37:26] DEBUG[18498]: chan_sip.c:3821 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '72414b7f092963a84f81480156e67ecd@91.216.53.28:5060' Request 102: Found [2011-01-14 10:37:26] DEBUG[18498]: chan_sip.c:3089 __sip_xmit: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.1.16.97:2414 [2011-01-14 10:37:26] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'Local/9d25user1@default-fd9b;1' [2011-01-14 10:37:26] DEBUG[18498]: chan_local.c:864 local_hangup: This local call has the ANSWERED_ELSEWHERE flag set. [2011-01-14 10:37:26] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format alaw [2011-01-14 10:37:26] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:26] DEBUG[18498]: app_queue.c:2611 leave_queue: Queue 'IVR-52-1' Leave, Channel 'SIP/ngn54-00005953' [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 2137065805 to 473495185 due to a source change [2011-01-14 10:37:26] DEBUG[18498]: channel.c:6807 ast_generic_bridge: Got a FRAME_CONTROL (-1) frame on channel Local/9d25user3@default-c1ca;1 [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:26] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and Local/9d25user3@default-c1ca;1 [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 473495185 to 1462623774 due to a source change [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1462623774 to 753868907 due to a source change [2011-01-14 10:37:26] DEBUG[18498]: channel.c:6426 ast_do_masquerade: Putting channel SIP/user3-00005966 in alaw/alaw formats [2011-01-14 10:37:26] DEBUG[18498]: chan_sip.c:6172 sip_fixup: SIP Fixup: New owner for dialogue 53e78e6e2e4f02f7779127095fb8d553@91.216.53.28:5060: SIP/user3-00005966 (Old parent: Local/9d25user3@default-c1ca;1) [2011-01-14 10:37:26] DEBUG[18498]: channel.c:6471 ast_do_masquerade: Released clone lock on 'Local/9d25user3@default-c1ca;1' [2011-01-14 10:37:26] DEBUG[18498]: channel.c:6479 ast_do_masquerade: Done Masquerading SIP/user3-00005966 (6) [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 753868907 to 328506215 due to a source change [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:737 ast_rtp_change_source: Not changing SSRC since we haven't sent any RTP yet [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:1239 ast_rtp_write: Ooh, format changed from unknown to alaw [2011-01-14 10:37:26] DEBUG[18498]: res_rtp_asterisk.c:1270 ast_rtp_write: Created smoother: format: alaw ms: 20 len: 160 [2011-01-14 10:37:27] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:28] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:29] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:29] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 176 bytes [2011-01-14 10:37:30] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:31] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:32] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:32] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 176 bytes [2011-01-14 10:37:33] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:34] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:35] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:35] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 176 bytes [2011-01-14 10:37:37] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:37] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:38] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:39] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 176 bytes [2011-01-14 10:37:39] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:40] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:41] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:41] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:42] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 176 bytes [2011-01-14 10:37:42] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/user3-00005966 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF begin on channel (SIP/user3-00005966) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 328506215 to 2038946136 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1954149760 to 1944021543 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/user3-00005966, duration 120 ms [2011-01-14 10:37:44] DTMF[18498]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF end on channel (SIP/user3-00005966) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: features.c:2674 feature_interpret: Feature interpret: chan=SIP/ngn54-00005953, peer=SIP/user3-00005966, code=*, sense=2, features=2, dynamic=# [2011-01-14 10:37:44] DEBUG[18498]: features.c:3414 ast_bridge_call: Set feature timer to 2000 [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 2038946136 to 1804017141 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1944021543 to 1976092474 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/user3-00005966 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF begin on channel (SIP/user3-00005966) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1804017141 to 906989512 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1976092474 to 1688042700 due to a source change [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/user3-00005966, duration 120 ms [2011-01-14 10:37:44] DTMF[18498]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF end on channel (SIP/user3-00005966) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:44] DEBUG[18498]: features.c:2674 feature_interpret: Feature interpret: chan=SIP/ngn54-00005953, peer=SIP/user3-00005966, code=**, sense=2, features=2, dynamic=# [2011-01-14 10:37:44] DEBUG[18498]: features.c:2566 feature_interpret_helper: Feature detected: fname=Attended Transfer sname=atxfer exten=** [2011-01-14 10:37:44] DEBUG[18498]: features.c:1837 builtin_atxfer: Executing Attended Transfer SIP/ngn54-00005953, SIP/user3-00005966 (sense=2) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:44] VERBOSE[18498]: res_musiconhold.c:426 moh_files_alloc: [2011-01-14 10:37:44] -- Started music on hold, class 'default', on SIP/ngn54-00005953 [2011-01-14 10:37:44] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:44] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format gsm [2011-01-14 10:37:44] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:44] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:37:44] -- Playing 'pbx-transfer.gsm' (language 'ru') [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:44] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:45] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 248 bytes [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:45] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:45] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format slin [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3492 ast_read_generator_actions: Generator got voice, switching to phase locked mode [2011-01-14 10:37:45] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/user3-00005966 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/user3-00005966 [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/user3-00005966, duration 120 ms [2011-01-14 10:37:46] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/user3-00005966 [2011-01-14 10:37:46] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/user3-00005966 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/user3-00005966 [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/user3-00005966, duration 100 ms [2011-01-14 10:37:46] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/user3-00005966 [2011-01-14 10:37:46] WARNING[18498]: features.c:1861 builtin_atxfer: Did not read data. [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:46] VERBOSE[18498]: res_musiconhold.c:267 moh_files_release: [2011-01-14 10:37:46] -- Stopped music on hold on SIP/ngn54-00005953 [2011-01-14 10:37:46] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format alaw [2011-01-14 10:37:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:46] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format gsm [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1061 ast_rtp_raw_write: Difference is 3920, ms is 510 [2011-01-14 10:37:46] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:46] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:37:46] -- Playing 'beeperr.gsm' (language 'ru') [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/user3-00005966 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/user3-00005966 [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:46] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/user3-00005966, duration 140 ms [2011-01-14 10:37:46] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/user3-00005966 [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:46] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:47] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:47] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:47] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:47] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:47] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 906989512 to 243991824 due to a source change [2011-01-14 10:37:47] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1688042700 to 8251086 due to a source change [2011-01-14 10:37:47] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 272 bytes [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:48] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/user3-00005966 [2011-01-14 10:37:48] DTMF[18498]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:48] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF begin on channel (SIP/user3-00005966) [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:48] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 243991824 to 2025856536 due to a source change [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 8251086 to 465127068 due to a source change [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:48] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/user3-00005966, duration 140 ms [2011-01-14 10:37:49] DTMF[18498]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/user3-00005966 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF end on channel (SIP/user3-00005966) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: features.c:2674 feature_interpret: Feature interpret: chan=SIP/ngn54-00005953, peer=SIP/user3-00005966, code=*, sense=2, features=2, dynamic=# [2011-01-14 10:37:49] DEBUG[18498]: features.c:3414 ast_bridge_call: Set feature timer to 2000 [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 2025856536 to 1075525920 due to a source change [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 465127068 to 1828334333 due to a source change [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/user3-00005966 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF begin on channel (SIP/user3-00005966) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1075525920 to 2054700344 due to a source change [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1828334333 to 1709257974 due to a source change [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 42 (*), at 10.1.16.22:54906 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/user3-00005966, duration 200 ms [2011-01-14 10:37:49] DTMF[18498]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/user3-00005966 [2011-01-14 10:37:49] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: channel.c:6831 ast_generic_bridge: Got DTMF end on channel (SIP/user3-00005966) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:37:49] DEBUG[18498]: features.c:2674 feature_interpret: Feature interpret: chan=SIP/ngn54-00005953, peer=SIP/user3-00005966, code=**, sense=2, features=2, dynamic=# [2011-01-14 10:37:49] DEBUG[18498]: features.c:2566 feature_interpret_helper: Feature detected: fname=Attended Transfer sname=atxfer exten=** [2011-01-14 10:37:49] DEBUG[18498]: features.c:1837 builtin_atxfer: Executing Attended Transfer SIP/ngn54-00005953, SIP/user3-00005966 (sense=2) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:49] VERBOSE[18498]: res_musiconhold.c:426 moh_files_alloc: [2011-01-14 10:37:49] -- Started music on hold, class 'default', on SIP/ngn54-00005953 [2011-01-14 10:37:49] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:49] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format gsm [2011-01-14 10:37:49] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:49] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:37:49] -- Playing 'pbx-transfer.gsm' (language 'ru') [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:49] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 0000000a (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:50] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format slin [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3492 ast_read_generator_actions: Generator got voice, switching to phase locked mode [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 10.1.16.22:54906 [2011-01-14 10:37:50] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/user3-00005966 [2011-01-14 10:37:50] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/user3-00005966 [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 49 (1), at 10.1.16.22:54906 [2011-01-14 10:37:50] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/user3-00005966, duration 160 ms [2011-01-14 10:37:50] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/user3-00005966 [2011-01-14 10:37:50] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:50] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:50] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000001 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/user3-00005966 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/user3-00005966 [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 344 bytes [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/user3-00005966, duration 140 ms [2011-01-14 10:37:51] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/user3-00005966 [2011-01-14 10:37:51] WARNING[18498]: features.c:1861 builtin_atxfer: Did not read data. [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:37:51] VERBOSE[18498]: res_musiconhold.c:267 moh_files_release: [2011-01-14 10:37:51] -- Stopped music on hold on SIP/ngn54-00005953 [2011-01-14 10:37:51] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/ngn54-00005953 to write format alaw [2011-01-14 10:37:51] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:51] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format gsm [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1061 ast_rtp_raw_write: Difference is 3584, ms is 468 [2011-01-14 10:37:51] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [2011-01-14 10:37:51] VERBOSE[18498]: file.c:983 ast_streamfile: [2011-01-14 10:37:51] -- Playing 'beeperr.gsm' (language 'ru') [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/user3-00005966 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/user3-00005966 [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1378 create_dtmf_frame: Sending dtmf: 51 (3), at 10.1.16.22:54906 [2011-01-14 10:37:51] DTMF[18498]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/user3-00005966, duration 80 ms [2011-01-14 10:37:51] DTMF[18498]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/user3-00005966 [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1422 process_dtmf_rfc2833: - RTP 2833 Event: 00000003 (len = 4) [2011-01-14 10:37:51] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:51] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:51] DEBUG[18498]: channel.c:3375 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [2011-01-14 10:37:51] DEBUG[18498]: channel.c:4963 set_format: Set channel SIP/user3-00005966 to write format alaw [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 2054700344 to 218412123 due to a source change [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:744 ast_rtp_change_source: Changing ssrc from 1709257974 to 2017419298 due to a source change [2011-01-14 10:37:51] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:52] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:53] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:54] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:37:54] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:55] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:55] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:56] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:57] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:37:57] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:58] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:37:59] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:00] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:38:00] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:01] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:02] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:02] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:03] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:38:04] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:04] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:05] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:06] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:38:07] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:07] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:08] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:09] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:38:09] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:10] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:11] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:12] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 368 bytes [2011-01-14 10:38:12] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:14] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:15] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:15] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 344 bytes [2011-01-14 10:38:16] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:16] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:18] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:18] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:19] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:20] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:21] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:22] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:23] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:24] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:24] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:25] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:25] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:26] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:27] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:28] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:29] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:29] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:30] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:30] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:31] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:32] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:33] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:34] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:35] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:36] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:36] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 320 bytes [2011-01-14 10:38:37] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:38] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 80 bytes [2011-01-14 10:38:38] DEBUG[18498]: res_rtp_asterisk.c:1673 ast_rtcp_read: Got RTCP report of 160 bytes [2011-01-14 10:38:38] DEBUG[18498]: channel.c:6766 ast_generic_bridge: Didn't get a frame from channel: SIP/user3-00005966 [2011-01-14 10:38:38] DEBUG[18498]: res_rtp_asterisk.c:725 ast_rtp_update_source: Setting the marker bit due to a source update [2011-01-14 10:38:38] DEBUG[18498]: channel.c:7224 ast_channel_bridge: Bridge stops bridging channels SIP/ngn54-00005953 and SIP/user3-00005966 [2011-01-14 10:38:38] DEBUG[18498]: cdr_mysql.c:334 mysql_log: Inserting a CDR record. [2011-01-14 10:38:38] DEBUG[18498]: cdr_mysql.c:337 mysql_log: SQL command as follows: INSERT INTO cdr (calldate,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES ('2011-01-14 10:36:32','9127778888','1','IVR_1234552','SIP/ngn54-00005953','Local/9d25user3@default-c1ca;1','Queue','IVR-52-1,trR,,,1800','126','126','ANSWERED','3','1294983392.33084') [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '9127778888' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"9127778888"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '9127778888' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"9127778888"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '1' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"1"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'IVR_1234552' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"IVR_1234552"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'SIP/ngn54-00005953' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"SIP/ngn54-00005953"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'Local/9d25user3@default-c1ca;1' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"Local/9d25user3@default-c1ca;1"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'Queue' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"Queue"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'IVR-52-1,trR,,,1800' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"IVR-52-1,trR,,,1800"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '2011-01-14 10:36:32' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"2011-01-14 10:36:32"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '2011-01-14 10:36:32' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"2011-01-14 10:36:32"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '2011-01-14 10:38:38' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"2011-01-14 10:38:38"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '126' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"126"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '126' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"126"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'ANSWERED' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"ANSWERED"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is 'DOCUMENTATION' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"DOCUMENTATION"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '(null)' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '""' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '1294983392.33084' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '"1294983392.33084"' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '(null)' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '""' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:3696 ast_str_substitute_variables_full: Function result is '35853' [2011-01-14 10:38:38] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'SIP/user3-00005966' [2011-01-14 10:38:38] DEBUG[18498]: chan_sip.c:5819 sip_hangup: Hangup call SIP/user3-00005966, SIP callid 53e78e6e2e4f02f7779127095fb8d553@91.216.53.28:5060 [2011-01-14 10:38:38] DEBUG[18498]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2f864c8' [2011-01-14 10:38:38] DEBUG[18498]: pbx.c:4752 __ast_pbx_run: Spawn extension (IVR_1234552,1,1) exited non-zero on 'SIP/ngn54-00005953' [2011-01-14 10:38:38] VERBOSE[18498]: pbx.c:4753 __ast_pbx_run: [2011-01-14 10:38:38] == Spawn extension (IVR_1234552, 1, 1) exited non-zero on 'SIP/ngn54-00005953' [2011-01-14 10:38:38] DEBUG[18498]: channel.c:2578 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/ngn54-00005953' [2011-01-14 10:38:38] DEBUG[18498]: channel.c:2706 ast_hangup: Hanging up channel 'SIP/ngn54-00005953' [2011-01-14 10:38:38] DEBUG[18498]: chan_sip.c:5819 sip_hangup: Hangup call SIP/ngn54-00005953, SIP callid c5d66038d29596a220e85ee69a5b77c171452b@62.148.237.132 [2011-01-14 10:38:38] DEBUG[18498]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fca702fe968' [2011-01-14 10:38:38] DEBUG[18498]: chan_sip.c:3089 __sip_xmit: Trying to put 'BYE sip:62.' onto UDP socket destined for 62.148.237.132:5060