Есть астериск в виде Elastix-а на сервере, подключен к роутеру. В этот же роутер заведена линия интернета и GSM-шлюз GoIP 4.
Настроил исходящие, при звонке с софт-звонилки (Zoiper настроен в качестве Extension-а) звонок уходит в транк и все хорошо. Звонок перекидывается на GSM-шлюз и вызов отлично приходит на мобильный телефон. Но если провайдер делает прозвон (до провайдера прокинуты sip-транки), то приходит вызов, уходит в транк до шлюза и вот тут начинается что то неясное. Судя по логам вызов доходит до шлюза. В WEB-интерфейсе гоипа висит статус DIALING:XXXXXXXXX. Провайдер говорит что на его стороне в телефоне идут гудки.
Но при этом на мобильный телефон вызов не падает.
Вот так же лог того что происходит:
Код: Выделить всё
[2015-06-12 11:19:19] DEBUG[2638]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 1b0fd76810e411e59baa6c3be551a83c@Айпи провайдера - INVITE (No RTP)
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2b2e90029238'
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: res_rtp_asterisk.c:2415 ast_rtp_new: Allocated port ПОРТ for RTP instance '0x2b2e90029238'
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x2b2e90029238' is setup and ready to go
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: res_rtp_asterisk.c:4544 ast_rtp_prop_set: Setup RTCP on RTP instance '0x2b2e90029238'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: chan_sip.c:5586 do_setnat: Setting NAT on RTP to Off
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x2b2e8d198530
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x2b2e8d198530
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x2b2e8d198530
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: res_rtp_asterisk.c:4599 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2b2e90029238'
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: res_rtp_asterisk.c:4510 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x2b2e90029238'
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: chan_sip.c:25580 handle_request_invite: Checking SIP call limits for device Provider-2
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'NoOp'
-- Executing [номер на который звонят@from-trunk:1] NoOp("SIP/Provider-2-00000052", "Catch-All DID Match - Found номер на который звонят - You probably want a DID for this.") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [номер на который звонят7@from-trunk:2] Set("SIP/Provider-2-00000052", "__FROM_DID=номер на который звонят") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Goto'
-- Executing [номер на который звонят@from-trunk:3] Goto("SIP/Provider-2-00000052", "ext-did,s,1") in new stack
-- Goto (ext-did,s,1)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '0'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'ExecIf'
-- Executing [s@ext-did:1] ExecIf("SIP/Provider-2-00000052", "0?Set(__FROM_DID=s)") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Gosub'
-- Executing [s@ext-did:2] Gosub("SIP/Provider-2-00000052", "app-blacklist-check,s,1()") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: app_stack.c:579 gosub_exec: Channel SIP/Provider-2-00000052 has no datastore, so we're allocating one.
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: db.c:376 db_get_common: Unable to find key 'номер с которого звонит провайдер ' in family 'blacklist'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: db.c:376 db_get_common: Unable to find key '' in family 'blacklist'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '0'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'GotoIf'
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/Provider-2-00000052", "0?blacklisted") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:11836 pbx_builtin_gotoif: Not taking any branch
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@app-blacklist-check:2] Set("SIP/Provider-2-00000052", "CALLED_BLACKLIST=1") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Return'
-- Executing [s@app-blacklist-check:3] Return("SIP/Provider-2-00000052", "") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@ext-did:3] Set("SIP/Provider-2-00000052", "CDR(did)=номер на который звонят") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '1'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'ExecIf'
-- Executing [s@ext-did:4] ExecIf("SIP/Provider-2-00000052", "1 ?Set(CALLERID(name)=номер с которого звонит провайдер)") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@ext-did:5] Set("SIP/Provider-2-00000052", "CHANNEL(musicclass)=default") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@ext-did:6] Set("SIP/Provider-2-00000052", "__MOHCLASS=default") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@ext-did:7] Set("SIP/Provider-2-00000052", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [s@ext-did:8] Set("SIP/Provider-2-00000052", "CALLERPRES()=allowed_not_screened") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Goto'
-- Executing [s@ext-did:9] Goto("SIP/Provider-2-00000052", "ext-trunk,1,1") in new stack
-- Goto (ext-trunk,1,1)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [1@ext-trunk:1] Set("SIP/Provider-2-00000052", "TDIAL_STRING=SIP/9000") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [1@ext-trunk:2] Set("SIP/Provider-2-00000052", "DIAL_TRUNK=1") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Goto'
-- Executing [1@ext-trunk:3] Goto("SIP/Provider-2-00000052", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:1] Set("SIP/Provider-2-00000052", "OUTBOUND_GROUP=OUT_1") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '1'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'GotoIf'
-- Executing [tdial@ext-trunk:2] GotoIf("SIP/Provider-2-00000052", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '1'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'ExecIf'
-- Executing [tdial@ext-trunk:4] ExecIf("SIP/Provider-2-00000052", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:5] Set("SIP/Provider-2-00000052", "DIAL_NUMBER=номер на который звонят") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '0'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'GosubIf'
-- Executing [tdial@ext-trunk:6] GosubIf("SIP/Provider-2-00000052", "0?sub-flp-1,s,1()") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:7] Set("SIP/Provider-2-00000052", "OUTNUM=10номер на который звонят") in new stack // 10 это префикс который выдает транк на выходе
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: db.c:376 db_get_common: Unable to find key '1/dialopts' in family 'TRUNK'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4783 pbx_substitute_variables_helper_full: Expression result is '0'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:8] Set("SIP/Provider-2-00000052", "DIAL_TRUNK_OPTIONS=") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Dial'
-- Executing [tdial@ext-trunk:9] Dial("SIP/Provider-2-00000052", "SIP/9000/10номер на который звонят,300,") in new stack
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:29758 sip_request_call: Asked to create a SIP channel with formats: (ulaw)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 6147dd9f0f12ab433ae82e501d218098@127.0.0.1:5060 - INVITE (No RTP)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2cf22f8'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: res_rtp_asterisk.c:2415 ast_rtp_new: Allocated port 17112 for RTP instance '0x2cf22f8'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x2cf22f8' is setup and ready to go
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: res_rtp_asterisk.c:4544 ast_rtp_prop_set: Setup RTCP on RTP instance '0x2cf22f8'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:5586 do_setnat: Setting NAT on RTP to Off
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:5586 do_setnat: Setting NAT on RTP to Off
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:8427 change_callid_pvt: SIP call-id changed from '6147dd9f0f12ab433ae82e501d218098@127.0.0.1:5060' to '2e015bf178739f6a550688843355e338@192.168.0.133:5060'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: rtp_engine.c:1714 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/9000-00000053' with that of 'SIP/Provider-2-00000052'
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: channel.c:6549 ast_channel_inherit_variables: Inheriting variable CALLINGPRES_SV from SIP/Provider-2-00000052 to SIP/9000-00000053.
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: channel.c:6549 ast_channel_inherit_variables: Inheriting variable MOHCLASS from SIP/Provider-2-00000052 to SIP/9000-00000053.
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: channel.c:6549 ast_channel_inherit_variables: Inheriting variable FROM_DID from SIP/Provider-2-00000052 to SIP/9000-00000053.
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:6207 sip_call: Outgoing Call for 10номер на который звонят
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:13085 add_sdp: ** Our capability: (gsm|ulaw|alaw) Video flag: False Text flag: False
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:13086 add_sdp: ** Our prefcodec: (ulaw)
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method INVITE - callid 2e015bf178739f6a550688843355e338@192.168.0.133:5060
-- Called SIP/9000/10номер на который звонят
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: chan_sip.c:4457 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2e015bf178739f6a550688843355e338@192.168.0.133:5060' Request 102: Found
[2015-06-12 11:19:19] DEBUG[2638][C-00000029]: chan_sip.c:4457 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2e015bf178739f6a550688843355e338@192.168.0.133:5060' Request 102: Found
-- SIP/9000-00000053 is ringing
[2015-06-12 11:19:19] DEBUG[10924][C-00000029]: rtp_engine.c:1821 ast_rtp_instance_early_bridge: Setting early bridge SDP of 'SIP/Provider-2-00000052' with that of 'SIP/9000-00000053'
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 35a8961b439c632d09045abf2561ff0d@127.0.0.1:5060 - OPTIONS (No RTP)
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8427 change_callid_pvt: SIP call-id changed from '35a8961b439c632d09045abf2561ff0d@127.0.0.1:5060' to '1b1bc4074fcce05505c97495287c426e@192.168.0.133:5060'
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method OPTIONS - callid 1b1bc4074fcce05505c97495287c426e@192.168.0.133:5060
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '1b1bc4074fcce05505c97495287c426e@192.168.0.133:5060' of Request 102: Match Found
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 455a38ae55c9952c2c68ea631127991b@127.0.0.1:5060 - OPTIONS (No RTP)
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8427 change_callid_pvt: SIP call-id changed from '455a38ae55c9952c2c68ea631127991b@127.0.0.1:5060' to '246dcb936a33ef972f912d767701f252@192.168.0.133:5060'
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method OPTIONS - callid 246dcb936a33ef972f912d767701f252@192.168.0.133:5060
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 4cedad0c64576c091814239f0bcb5855@127.0.0.1:5060 - OPTIONS (No RTP)
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8427 change_callid_pvt: SIP call-id changed from '4cedad0c64576c091814239f0bcb5855@127.0.0.1:5060' to '0a8e0d3e4b6bc78957b9631d3678f547@192.168.0.133:5060'
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method OPTIONS - callid 0a8e0d3e4b6bc78957b9631d3678f547@192.168.0.133:5060
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '246dcb936a33ef972f912d767701f252@192.168.0.133:5060' of Request 102: Match Found
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '0a8e0d3e4b6bc78957b9631d3678f547@192.168.0.133:5060' of Request 102: Match Found
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8638 sip_alloc: Allocating new SIP dialog for 1535b987254ca8976b4e1bc156ef906c@127.0.0.1:5060 - OPTIONS (No RTP)
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:8427 change_callid_pvt: SIP call-id changed from '1535b987254ca8976b4e1bc156ef906c@127.0.0.1:5060' to '705fbbb840de33975a3cdc7c22c3a6fa@192.168.0.133:5060'
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:3367 initialize_initreq: Initializing initreq for method OPTIONS - callid 705fbbb840de33975a3cdc7c22c3a6fa@192.168.0.133:5060
[2015-06-12 11:19:27] DEBUG[2638]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '705fbbb840de33975a3cdc7c22c3a6fa@192.168.0.133:5060' of Request 102: Match Found
[2015-06-12 11:19:59] DEBUG[2638][C-00000029]: chan_sip.c:4378 __sip_ack: Acked pending invite 102
[2015-06-12 11:19:59] DEBUG[2638][C-00000029]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '2e015bf178739f6a550688843355e338@192.168.0.133:5060' of Request 102: Match Found
-- Got SIP response 503 "Service Unavailable" back from 192.168.0.134:5160 // 503 со шлюза приходит уже когда провайдер кладет трубку
[2015-06-12 11:19:59] DEBUG[2638][C-00000029]: res_rtp_asterisk.c:4599 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2cf22f8'
-- SIP/9000-00000053 is circuit-busy
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/9000-00000053'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: chan_sip.c:6914 sip_hangup: Hangup call SIP/9000-00000053, SIP callid 2e015bf178739f6a550688843355e338@192.168.0.133:5060
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: res_rtp_asterisk.c:4599 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2cf22f8'
== Everyone is busy/congested at this time (1:0/1/0)
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: app_dial.c:3102 dial_exec_full: Exiting with DIALSTATUS=CONGESTION.
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:10] Set("SIP/Provider-2-00000052", "CALLERID(number)=номер с которого звонит провайдер") in new stack
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Set'
-- Executing [tdial@ext-trunk:11] Set("SIP/Provider-2-00000052", "CALLERID(name)=номер с которого звонит провайдер") in new stack
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: pbx.c:4883 pbx_extension_helper: Launching 'Hangup'
-- Executing [tdial@ext-trunk:12] Hangup("SIP/Provider-2-00000052", "") in new stack
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/Provider-2-00000052'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: pbx.c:6572 __ast_pbx_run: Spawn extension (ext-trunk,tdial,12) exited non-zero on 'SIP/Provider-2-00000052'
== Spawn extension (ext-trunk, tdial, 12) exited non-zero on 'SIP/Provider-2-00000052'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: channel.c:2661 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/Provider-2-00000052'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: channel.c:2840 ast_hangup: Hanging up channel 'SIP/Provider-2-00000052'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: chan_sip.c:6914 sip_hangup: Hangup call SIP/Provider-2-00000052, SIP callid 1b0fd76810e411e59baa6c3be551a83c@айпи провайдера
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: res_rtp_asterisk.c:4599 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x2b2e90029238'
[2015-06-12 11:19:59] DEBUG[2638]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x2cf22f8'
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: cdr_mysql.c:336 mysql_log: Inserting a CDR record.
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: cdr_mysql.c:339 mysql_log: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`dstchannel`,`lastapp`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`) VALUES ('2015-06-12 11:19:19','\"номер с которого звонит провайдер4\" <номер с которого звонит провайдер>','номер с которого звонит провайдер','tdial','ext-trunk','SIP/Provider-2-00000052','SIP/9000-00000053','Hangup','40','0','FAILED','3','1434100759.82')
[2015-06-12 11:19:59] DEBUG[10924][C-00000029]: cdr_sqlite3_custom.c:261 write_cdr: About to log: INSERT INTO cdr (calldate,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,test,src,dst) VALUES ('2015-06-12 11:19:19','"номер с которого звонит провайдер" <номер с которого звонит провайдер>','ext-trunk','SIP/Provider-2-00000052','SIP/9000-00000053','Hangup','','40','0','FAILED','DOCUMENTATION','','1434100759.82','','','номер с которого звонит провайдер','tdial')
[2015-06-12 11:19:59] DEBUG[2638][C-00000029]: chan_sip.c:4416 __sip_ack: Stopping retransmission on '1b0fd76810e411e59baa6c3be551a83c@айпи провайдера' of Response 1: Match Found
-- Remote UNIX connection
-- Remote UNIX connection disconnected
[2015-06-12 11:20:05] DEBUG[2638]: rtp_engine.c:226 instance_destructor: Destroyed RTP instance '0x2b2e90029238'
Всем спасибо за помощь.