Странная проблема с исходящими вызовами GoIP4

Sparkle · Сообщение **Sparkle** » 16 июн 2015, 13:43

Здравствуйте!
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'

Вы уж извините что все так подробно вплоть до роутера, но лучше изначально все расписать=)
Всем спасибо за помощь.

ded · Сообщение **ded** » 16 июн 2015, 16:35

Dial("SIP/Provider-2-00000052", "SIP/9000/10номер на который звонят,300,") - почему 10номер на который звонят? Откуда 10?
Вы же по Украине звоните с ноля? Должно было быть что-то типа SIP/9000/0123456789 (исходя из того, что "SIP/9000 - это GoIP)?
Или вы по префиксу 10 на GoIP разруливаете в какую симку терминировать?

Так и непонятны диалоги на localhost:
Allocating new SIP dialog for 455a38ae55c9952c2c68ea631127991b@127.0.0.1:5060

Sparkle · Сообщение **Sparkle** » 16 июн 2015, 16:38

Да. Префикс выдается в транке. На шлюз приходит 10 бла бла бла и там в зависимости от префикса выбирается симслот. Ну и отсекается при наборе=)

Sparkle · Сообщение **Sparkle** » 16 июн 2015, 16:42

10 это префикс для выбора симслота на шлюзе.
А вот откуда диалоги на 127.0.0.1 вообще понятия не имею.

ded · Сообщение **ded** » 17 июн 2015, 00:27

Два тестовых звонка:
1) с софтфона на 10бла бла бла
и при этом посмотреть
CLI> sip show peers

2) входящий от provider-2 на GoIP 10бла бла бла
и при этом посмотреть
CLI> sip show peers

и сравнить две картинки. Если не понятно - то повторить со включенными опциями
CLI> sip set debug on

Sparkle · Сообщение **Sparkle** » 17 июн 2015, 13:44

Посмотрел логи при вызовах, ничего не увидел=(

Вызов с софтфона в локальной сети:

Код: Выделить всё

 -- Executing [Номер на который звонили@from-trunk:1] NoOp("SIP/8000-00000026", "Catch-All DID Match - Found Номер на который звонили - You probably want a DID for this.") in new stack
    -- Executing [Номер на который звонили@from-trunk:2] Set("SIP/8000-00000026", "__FROM_DID=Номер на который звонили") in new stack
    -- Executing [Номер на который звонили@from-trunk:3] Goto("SIP/8000-00000026", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] ExecIf("SIP/8000-00000026", "0?Set(__FROM_DID=s)") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/8000-00000026", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/8000-00000026", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/8000-00000026", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/8000-00000026", "") in new stack
    -- Executing [s@ext-did:3] Set("SIP/8000-00000026", "CDR(did)=Номер на который звонили") in new stack
    -- Executing [s@ext-did:4] ExecIf("SIP/8000-00000026", "0 ?Set(CALLERID(name)=8000)") in new stack
    -- Executing [s@ext-did:5] Set("SIP/8000-00000026", "CHANNEL(musicclass)=default") in new stack
    -- Executing [s@ext-did:6] Set("SIP/8000-00000026", "__MOHCLASS=default") in new stack
    -- Executing [s@ext-did:7] Set("SIP/8000-00000026", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:8] Set("SIP/8000-00000026", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [s@ext-did:9] Goto("SIP/8000-00000026", "ext-trunk,1,1") in new stack
    -- Goto (ext-trunk,1,1)
    -- Executing [1@ext-trunk:1] Set("SIP/8000-00000026", "TDIAL_STRING=SIP/9000") in new stack
    -- Executing [1@ext-trunk:2] Set("SIP/8000-00000026", "DIAL_TRUNK=1") in new stack
    -- Executing [1@ext-trunk:3] Goto("SIP/8000-00000026", "ext-trunk,tdial,1") in new stack
    -- Goto (ext-trunk,tdial,1)
    -- Executing [tdial@ext-trunk:1] Set("SIP/8000-00000026", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [tdial@ext-trunk:2] GotoIf("SIP/8000-00000026", "1?nomax") in new stack
    -- Goto (ext-trunk,tdial,4)
    -- Executing [tdial@ext-trunk:4] ExecIf("SIP/8000-00000026", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
    -- Executing [tdial@ext-trunk:5] Set("SIP/8000-00000026", "DIAL_NUMBER=Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:6] GosubIf("SIP/8000-00000026", "0?sub-flp-1,s,1()") in new stack
    -- Executing [tdial@ext-trunk:7] Set("SIP/8000-00000026", "OUTNUM=10Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:8] Set("SIP/8000-00000026", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [tdial@ext-trunk:9] Dial("SIP/8000-00000026", "SIP/9000/10Номер на который звонили,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/9000/10Номер на который звонили
    -- SIP/9000-00000027 is ringing
    -- SIP/9000-00000027 is making progress passing it to SIP/8000-00000026 // пошли гудки
    -- SIP/9000-00000027 is making progress passing it to SIP/8000-00000026 // пошли гудки
  == Spawn extension (ext-trunk, tdial, 9) exited non-zero on 'SIP/8000-00000026'  // сброс на мобильном телефоне

Вызов от провайдера:

Код: Выделить всё

-- Executing [Номер на который звонили@from-trunk:1] NoOp("SIP/Provider-4-0000002c", "Catch-All DID Match - Found Номер на который звонили - You probably want a DID for this.") in new stack
    -- Executing [Номер на который звонили@from-trunk:2] Set("SIP/Provider-4-0000002c", "__FROM_DID=Номер на который звонили") in new stack
    -- Executing [Номер на который звонили@from-trunk:3] Goto("SIP/Provider-4-0000002c", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] ExecIf("SIP/Provider-4-0000002c", "0?Set(__FROM_DID=s)") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/Provider-4-0000002c", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/Provider-4-0000002c", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/Provider-4-0000002c", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/Provider-4-0000002c", "") in new stack
    -- Executing [s@ext-did:3] Set("SIP/Provider-4-0000002c", "CDR(did)=Номер на который звонили") in new stack
    -- Executing [s@ext-did:4] ExecIf("SIP/Provider-4-0000002c", "1 ?Set(CALLERID(name)=Номер с которого звонили)") in new stack
    -- Executing [s@ext-did:5] Set("SIP/Provider-4-0000002c", "CHANNEL(musicclass)=default") in new stack
    -- Executing [s@ext-did:6] Set("SIP/Provider-4-0000002c", "__MOHCLASS=default") in new stack
    -- Executing [s@ext-did:7] Set("SIP/Provider-4-0000002c", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:8] Set("SIP/Provider-4-0000002c", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [s@ext-did:9] Goto("SIP/Provider-4-0000002c", "ext-trunk,1,1") in new stack
    -- Goto (ext-trunk,1,1)
    -- Executing [1@ext-trunk:1] Set("SIP/Provider-4-0000002c", "TDIAL_STRING=SIP/9000") in new stack
    -- Executing [1@ext-trunk:2] Set("SIP/Provider-4-0000002c", "DIAL_TRUNK=1") in new stack
    -- Executing [1@ext-trunk:3] Goto("SIP/Provider-4-0000002c", "ext-trunk,tdial,1") in new stack
    -- Goto (ext-trunk,tdial,1)
    -- Executing [tdial@ext-trunk:1] Set("SIP/Provider-4-0000002c", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [tdial@ext-trunk:2] GotoIf("SIP/Provider-4-0000002c", "1?nomax") in new stack
    -- Goto (ext-trunk,tdial,4)
    -- Executing [tdial@ext-trunk:4] ExecIf("SIP/Provider-4-0000002c", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
    -- Executing [tdial@ext-trunk:5] Set("SIP/Provider-4-0000002c", "DIAL_NUMBER=Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:6] GosubIf("SIP/Provider-4-0000002c", "0?sub-flp-1,s,1()") in new stack
    -- Executing [tdial@ext-trunk:7] Set("SIP/Provider-4-0000002c", "OUTNUM=10Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:8] Set("SIP/Provider-4-0000002c", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [tdial@ext-trunk:9] Dial("SIP/Provider-4-0000002c", "SIP/9000/10Номер на который звонили,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/9000/10Номер на который звонили
    -- SIP/9000-0000002d is ringing // ушло на шлюз
  == Spawn extension (ext-trunk, tdial, 9) exited non-zero on 'SIP/Provider-4-0000002c' // сброс

10 перед номером на который был звонок это префикс для разрута по симкам на шлюзе.
SIP/8000 - это экстеншн на котором зарегана софт звонилка.
SIP/9000 - это транк на котором GSM-шлюз (GoIP-4)

Sparkle · Сообщение **Sparkle** » 17 июн 2015, 14:35

Вот также дебаг от провайдера:

Код: Выделить всё

<--- SIP read from UDP:Айпи провайдера:5061 --->
INVITE sip:38Номер на который звонили@Наш внешний айпи;user=phone SIP/2.0
Via: SIP/2.0/UDP Айпи провайдера:5061;rport;branch=z9hG4bK-1585514903-3843154708-1496090241-2097126117
From: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>;tag=1512769943-3843154708-1496090241-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>
Call-ID: 97092b8214e311e581862c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Contact: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
Max-Forwards: 70
User-Agent: TS-v4.5.1-16aW
Cisco-Guid: 2533841968-350425573-2173054041-3852140412
Content-Length: 259

v=0
o=- 1434540342 1434540342 IN IP4 Провайдер.144
s=-
c=IN IP4 Провайдер.144
t=0 0
m=audio 30116 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (13 headers 12 lines) ---
Sending to Айпи провайдера:5061 (no NAT)
Sending to Айпи провайдера:5061 (no NAT)
Using INVITE request as basis request - 97092b8214e311e581862c59e59aff7c@Айпи провайдера
Found peer 'Provider-3' for ' Номер с которого звонили' from Айпи провайдера:5061
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port Провайдер.144:30116
Looking for 38Номер на который звонили in from-trunk (domain Наш внешний айпи)
list_route: hop: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>

<--- Transmitting (no NAT) to Айпи провайдера:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Айпи провайдера:5061;branch=z9hG4bK-1585514903-3843154708-1496090241-2097126117;received=Айпи провайдера;rport=5061
From: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>;tag=1512769943-3843154708-1496090241-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>
Call-ID: 97092b8214e311e581862c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:38Номер на который звонили@192.168.0.133:5060>
Content-Length: 0


<------------>
    -- Executing [38Номер на который звонили@from-trunk:1] NoOp("SIP/Provider-3-00000034", "Catch-All DID Match - Found 38Номер на который звонили - You probably want a DID for this.") in new stack
    -- Executing [38Номер на который звонили@from-trunk:2] Set("SIP/Provider-3-00000034", "__FROM_DID=38Номер на который звонили") in new stack
    -- Executing [38Номер на который звонили@from-trunk:3] Goto("SIP/Provider-3-00000034", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] ExecIf("SIP/Provider-3-00000034", "0?Set(__FROM_DID=s)") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/Provider-3-00000034", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/Provider-3-00000034", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/Provider-3-00000034", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/Provider-3-00000034", "") in new stack
    -- Executing [s@ext-did:3] Set("SIP/Provider-3-00000034", "CDR(did)=38Номер на который звонили") in new stack
    -- Executing [s@ext-did:4] ExecIf("SIP/Provider-3-00000034", "1 ?Set(CALLERID(name)= Номер с которого звонили)") in new stack
    -- Executing [s@ext-did:5] Set("SIP/Provider-3-00000034", "CHANNEL(musicclass)=default") in new stack
    -- Executing [s@ext-did:6] Set("SIP/Provider-3-00000034", "__MOHCLASS=default") in new stack
    -- Executing [s@ext-did:7] Set("SIP/Provider-3-00000034", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:8] Set("SIP/Provider-3-00000034", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [s@ext-did:9] Goto("SIP/Provider-3-00000034", "ext-trunk,1,1") in new stack
    -- Goto (ext-trunk,1,1)
    -- Executing [1@ext-trunk:1] Set("SIP/Provider-3-00000034", "TDIAL_STRING=SIP/9000") in new stack
    -- Executing [1@ext-trunk:2] Set("SIP/Provider-3-00000034", "DIAL_TRUNK=1") in new stack
    -- Executing [1@ext-trunk:3] Goto("SIP/Provider-3-00000034", "ext-trunk,tdial,1") in new stack
    -- Goto (ext-trunk,tdial,1)
    -- Executing [tdial@ext-trunk:1] Set("SIP/Provider-3-00000034", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [tdial@ext-trunk:2] GotoIf("SIP/Provider-3-00000034", "1?nomax") in new stack
    -- Goto (ext-trunk,tdial,4)
    -- Executing [tdial@ext-trunk:4] ExecIf("SIP/Provider-3-00000034", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
    -- Executing [tdial@ext-trunk:5] Set("SIP/Provider-3-00000034", "DIAL_NUMBER=38Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:6] GosubIf("SIP/Provider-3-00000034", "0?sub-flp-1,s,1()") in new stack
    -- Executing [tdial@ext-trunk:7] Set("SIP/Provider-3-00000034", "OUTNUM=1038Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:8] Set("SIP/Provider-3-00000034", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [tdial@ext-trunk:9] Dial("SIP/Provider-3-00000034", "SIP/9000/1038Номер на который звонили,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 17610
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.134:5160:
INVITE sip:1038Номер на который звонили@192.168.0.134:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK3f46078e
Max-Forwards: 70
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as2c1a3dcb
To: <sip:1038Номер на который звонили@192.168.0.134:5160>
Contact: <sip: Номер с которого звонили@192.168.0.133:5060>
Call-ID: 6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 2107062732 2107062732 IN IP4 192.168.0.133
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.0.133
t=0 0
m=audio 17610 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/9000/1038Номер на который звонили

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK3f46078e
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as2c1a3dcb
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1279100481
Call-ID: 6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060
CSeq: 102 INVITE
Contact: <sip:9000@192.168.0.134:5160>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK3f46078e
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as2c1a3dcb
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1279100481
Call-ID: 6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060
CSeq: 102 INVITE
Contact: <sip:9000@192.168.0.134:5160>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:9000@192.168.0.134:5160>
    -- SIP/9000-00000035 is ringing

<--- Transmitting (no NAT) to Айпи провайдера:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP Айпи провайдера:5061;branch=z9hG4bK-1585514903-3843154708-1496090241-2097126117;received=Айпи провайдера;rport=5061
From: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>;tag=1512769943-3843154708-1496090241-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as47edda88
Call-ID: 97092b8214e311e581862c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:38Номер на который звонили@192.168.0.133:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK3f46078e
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as2c1a3dcb
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1279100481
Call-ID: 6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060
CSeq: 102 INVITE
Contact: <sip:9000@192.168.0.134:5160>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Got SIP response 503 "Service Unavailable" back from 192.168.0.134:5160
set_destination: Parsing <sip:9000@192.168.0.134:5160> for address/port to send to
set_destination: set destination to 192.168.0.134:5160
Transmitting (no NAT) to 192.168.0.134:5160:
ACK sip:9000@192.168.0.134:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK3f46078e
Max-Forwards: 70
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as2c1a3dcb
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1279100481
Contact: <sip: Номер с которого звонили@192.168.0.133:5060>
Call-ID: 6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


---
    -- SIP/9000-00000035 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [tdial@ext-trunk:10] Set("SIP/Provider-3-00000034", "CALLERID(number)= Номер с которого звонили") in new stack
    -- Executing [tdial@ext-trunk:11] Set("SIP/Provider-3-00000034", "CALLERID(name)= Номер с которого звонили") in new stack
    -- Executing [tdial@ext-trunk:12] Hangup("SIP/Provider-3-00000034", "") in new stack
  == Spawn extension (ext-trunk, tdial, 12) exited non-zero on 'SIP/Provider-3-00000034'
Scheduling destruction of SIP dialog '97092b8214e311e581862c59e59aff7c@Айпи провайдера' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to Айпи провайдера:5061 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP Айпи провайдера:5061;branch=z9hG4bK-1585514903-3843154708-1496090241-2097126117;received=Айпи провайдера;rport=5061
From: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>;tag=1512769943-3843154708-1496090241-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as47edda88
Call-ID: 97092b8214e311e581862c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '6259d14038b7ce3d2693fc222fe0c209@192.168.0.133:5060' Method: INVITE

<--- SIP read from UDP:Айпи провайдера:5061 --->
ACK sip:38Номер на который звонили@Наш внешний айпи;user=phone SIP/2.0
Via: SIP/2.0/UDP Айпи провайдера:5061;rport;branch=z9hG4bK-1585514903-3843154708-1496090241-2097126117
From: <sip: Номер с которого звонили@Айпи провайдера:5061;user=phone>;tag=1512769943-3843154708-1496090241-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as47edda88
Call-ID: 97092b8214e311e581862c59e59aff7c@Айпи провайдера
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:Айпи провайдера:5062 --->
INVITE sip:38Номер на который звонили@Наш внешний айпи;user=phone SIP/2.0
Via: SIP/2.0/UDP Айпи провайдера:5062;rport;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Contact: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE
Max-Forwards: 70
User-Agent: TS-v4.5.1-16aW
Cisco-Guid: 2587626898-350425573-2174233689-3852140412
Content-Length: 259

v=0
o=- 1434540348 1434540348 IN IP4 Провайдер 
s=-
c=IN IP4 Провайдер 
t=0 0
m=audio 19858 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (13 headers 12 lines) ---
Sending to Айпи провайдера:5062 (no NAT)
Sending to Айпи провайдера:5062 (no NAT)
Using INVITE request as basis request - 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
Found peer 'Provider-1' for ' Номер с которого звонили' from Айпи провайдера:5062
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port Провайдер :19858
Looking for 38Номер на который звонили in from-trunk (domain Наш внешний айпи)
list_route: hop: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>

<--- Transmitting (no NAT) to Айпи провайдера:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP Айпи провайдера:5062;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117;received=Айпи провайдера;rport=5062
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:38Номер на который звонили@192.168.0.133:5060>
Content-Length: 0


<------------>
    -- Executing [38Номер на который звонили@from-trunk:1] NoOp("SIP/Provider-1-00000036", "Catch-All DID Match - Found 38Номер на который звонили - You probably want a DID for this.") in new stack
    -- Executing [38Номер на который звонили@from-trunk:2] Set("SIP/Provider-1-00000036", "__FROM_DID=38Номер на который звонили") in new stack
    -- Executing [38Номер на который звонили@from-trunk:3] Goto("SIP/Provider-1-00000036", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] ExecIf("SIP/Provider-1-00000036", "0?Set(__FROM_DID=s)") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/Provider-1-00000036", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/Provider-1-00000036", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("SIP/Provider-1-00000036", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/Provider-1-00000036", "") in new stack
    -- Executing [s@ext-did:3] Set("SIP/Provider-1-00000036", "CDR(did)=38Номер на который звонили") in new stack
    -- Executing [s@ext-did:4] ExecIf("SIP/Provider-1-00000036", "1 ?Set(CALLERID(name)= Номер с которого звонили)") in new stack
    -- Executing [s@ext-did:5] Set("SIP/Provider-1-00000036", "CHANNEL(musicclass)=default") in new stack
    -- Executing [s@ext-did:6] Set("SIP/Provider-1-00000036", "__MOHCLASS=default") in new stack
    -- Executing [s@ext-did:7] Set("SIP/Provider-1-00000036", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:8] Set("SIP/Provider-1-00000036", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [s@ext-did:9] Goto("SIP/Provider-1-00000036", "ext-trunk,1,1") in new stack
    -- Goto (ext-trunk,1,1)
    -- Executing [1@ext-trunk:1] Set("SIP/Provider-1-00000036", "TDIAL_STRING=SIP/9000") in new stack
    -- Executing [1@ext-trunk:2] Set("SIP/Provider-1-00000036", "DIAL_TRUNK=1") in new stack
    -- Executing [1@ext-trunk:3] Goto("SIP/Provider-1-00000036", "ext-trunk,tdial,1") in new stack
    -- Goto (ext-trunk,tdial,1)
    -- Executing [tdial@ext-trunk:1] Set("SIP/Provider-1-00000036", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [tdial@ext-trunk:2] GotoIf("SIP/Provider-1-00000036", "1?nomax") in new stack
    -- Goto (ext-trunk,tdial,4)
    -- Executing [tdial@ext-trunk:4] ExecIf("SIP/Provider-1-00000036", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
    -- Executing [tdial@ext-trunk:5] Set("SIP/Provider-1-00000036", "DIAL_NUMBER=38Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:6] GosubIf("SIP/Provider-1-00000036", "0?sub-flp-1,s,1()") in new stack
    -- Executing [tdial@ext-trunk:7] Set("SIP/Provider-1-00000036", "OUTNUM=1038Номер на который звонили") in new stack
    -- Executing [tdial@ext-trunk:8] Set("SIP/Provider-1-00000036", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [tdial@ext-trunk:9] Dial("SIP/Provider-1-00000036", "SIP/9000/1038Номер на который звонили,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11104
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.134:5160:
INVITE sip:1038Номер на который звонили@192.168.0.134:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
Max-Forwards: 70
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>
Contact: <sip: Номер с которого звонили@192.168.0.133:5060>
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 756053683 756053683 IN IP4 192.168.0.133
s=Asterisk PBX 11.13.0
c=IN IP4 192.168.0.133
t=0 0
m=audio 11104 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/9000/1038Номер на который звонили

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1073364917
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 INVITE
Contact: <sip:9000@192.168.0.134:5160>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1073364917
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 INVITE
Contact: <sip:9000@192.168.0.134:5160>
User-Agent: dble
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:9000@192.168.0.134:5160>
    -- SIP/9000-00000037 is ringing

<--- Transmitting (no NAT) to Айпи провайдера:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP Айпи провайдера:5062;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117;received=Айпи провайдера;rport=5062
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as7a68090b
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:38Номер на который звонили@192.168.0.133:5060>
Content-Length: 0


<------------>
Really destroying SIP dialog '97092b8214e311e581862c59e59aff7c@Айпи провайдера' Method: ACK
Reliably Transmitting (no NAT) to Провайдер.132:5061:
OPTIONS sip:Провайдер.132 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK639553c5
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as5caf6744
To: <sip:Провайдер.132>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 3f19ef43271b6bd34d94fdbf3bc89afb@192.168.0.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:Провайдер.132:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK639553c5;received=Наш внешний айпи
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as5caf6744
To: <sip:Провайдер.132>;tag=4176923803-3843154708-996978573-1017663973
Call-ID: 3f19ef43271b6bd34d94fdbf3bc89afb@192.168.0.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '3f19ef43271b6bd34d94fdbf3bc89afb@192.168.0.133:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to Провайдер.132:5062:
OPTIONS sip:Провайдер.132 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2474b6b9
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as165605d1
To: <sip:Провайдер.132>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 0a3e7f867f238f7a5cd5d3173b9d992b@192.168.0.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:Провайдер.132:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2474b6b9;received=Наш внешний айпи
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as165605d1
To: <sip:Провайдер.132>;tag=1107880603-3843154708-996978573-1017663973
Call-ID: 0a3e7f867f238f7a5cd5d3173b9d992b@192.168.0.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '0a3e7f867f238f7a5cd5d3173b9d992b@192.168.0.133:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to Айпи провайдера:5062:
OPTIONS sip:Айпи провайдера SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK1eac4d70
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as0a71eba1
To: <sip:Айпи провайдера>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 509c4e2c2bed63646d60715b07dd1fdf@192.168.0.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to Айпи провайдера:5061:
OPTIONS sip:Айпи провайдера SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK4712bd25
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as07198890
To: <sip:Айпи провайдера>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 2c5edda50dce5e0a781ac9a52f230e69@192.168.0.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:Айпи провайдера:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK1eac4d70;received=Наш внешний айпи
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as0a71eba1
To: <sip:Айпи провайдера>;tag=2893677468-3843154708-1496095361-2097126117
Call-ID: 509c4e2c2bed63646d60715b07dd1fdf@192.168.0.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '509c4e2c2bed63646d60715b07dd1fdf@192.168.0.133:5060' Method: OPTIONS

<--- SIP read from UDP:Айпи провайдера:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK4712bd25;received=Наш внешний айпи
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as07198890
To: <sip:Айпи провайдера>;tag=4269605788-3843154708-1496095361-2097126117
Call-ID: 2c5edda50dce5e0a781ac9a52f230e69@192.168.0.133:5060
CSeq: 102 OPTIONS
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, REGISTER, UPDATE
Accept: application/dtmf-relay
Accept: application/ISUP
Accept: application/media_control+xml
Accept: application/sdp
Supported: 100rel
Server: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '2c5edda50dce5e0a781ac9a52f230e69@192.168.0.133:5060' Method: OPTIONS
Really destroying SIP dialog '248998181@192.168.0.134' Method: REGISTER

<--- SIP read from UDP:192.168.0.132:59580 --->
REGISTER sip:192.168.0.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.132:59580;branch=z9hG4bK-d8754z-fa13ca320a03100c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>
To: "8000"<sip:8000@192.168.0.133:5060>
From: "8000"<sip:8000@192.168.0.133:5060>;tag=844cb34e
Call-ID: M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.
CSeq: 285 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.18815.0
Authorization: Digest username="8000",realm="asterisk",nonce="39c32191",uri="sip:192.168.0.133:5060",response="1c9d58f21a42f3f33564a6850daad48e",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.0.132:59580 (no NAT)
Sending to 192.168.0.132:59580 (no NAT)

<--- Transmitting (no NAT) to 192.168.0.132:59580 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.132:59580;branch=z9hG4bK-d8754z-fa13ca320a03100c-1---d8754z-;received=192.168.0.132;rport=59580
From: "8000"<sip:8000@192.168.0.133:5060>;tag=844cb34e
To: "8000"<sip:8000@192.168.0.133:5060>;tag=as17c6367b
Call-ID: M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.
CSeq: 285 REGISTER
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f443d72"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.0.132:59580 --->
REGISTER sip:192.168.0.133:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.132:59580;branch=z9hG4bK-d8754z-e155da299c5d5265-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>
To: "8000"<sip:8000@192.168.0.133:5060>
From: "8000"<sip:8000@192.168.0.133:5060>;tag=844cb34e
Call-ID: M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.
CSeq: 286 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.18815.0
Authorization: Digest username="8000",realm="asterisk",nonce="1f443d72",uri="sip:192.168.0.133:5060",response="0849bba505a6b8d320591eb35f3f4eb5",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.0.132:59580 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.132:59580:
OPTIONS sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK53c5dd92
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as2690a9b2
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 339109017365278a18f920ce03d73712@192.168.0.133:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.13.0)
Date: Wed, 17 Jun 2015 11:25:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.0.132:59580 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.132:59580;branch=z9hG4bK-d8754z-e155da299c5d5265-1---d8754z-;received=192.168.0.132;rport=59580
From: "8000"<sip:8000@192.168.0.133:5060>;tag=844cb34e
To: "8000"<sip:8000@192.168.0.133:5060>;tag=as17c6367b
Call-ID: M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.
CSeq: 286 REGISTER
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>;expires=120
Date: Wed, 17 Jun 2015 11:25:57 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '06507ce0125a89d472c143ef289473f1@192.168.0.133:5060' in 6848 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.132:59580:
NOTIFY sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK40eb51aa
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as27348241
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 06507ce0125a89d472c143ef289473f1@192.168.0.133:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.11.0(11.13.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.0.133
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog 'M2RmNzMwMWMzY2FmN2QwN2U1NjJmOTU0ODVmNjE0MTc.' in 32000 ms (Method: REGISTER)
Retransmitting #1 (no NAT) to 192.168.0.132:59580:
NOTIFY sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK40eb51aa
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.0.133>;tag=as27348241
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>
Contact: <sip:Unknown@192.168.0.133:5060>
Call-ID: 06507ce0125a89d472c143ef289473f1@192.168.0.133:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.11.0(11.13.0)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.0.133
Voice-Message: 0/0 (0/0)

---

<--- SIP read from UDP:192.168.0.132:59580 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK53c5dd92
Contact: <sip:192.168.0.132:59580>
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>;tag=013e0753
From: "Unknown"<sip:Unknown@192.168.0.133>;tag=as2690a9b2
Call-ID: 339109017365278a18f920ce03d73712@192.168.0.133:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '339109017365278a18f920ce03d73712@192.168.0.133:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.0.132:59580 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK40eb51aa
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>;tag=6d2b646a
From: "Unknown"<sip:Unknown@192.168.0.133>;tag=as27348241
Call-ID: 06507ce0125a89d472c143ef289473f1@192.168.0.133:5060
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.132:59580 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK40eb51aa
To: <sip:8000@192.168.0.132:59580;rinstance=a545132563f0a303>;tag=6d2b646a
From: "Unknown"<sip:Unknown@192.168.0.133>;tag=as27348241
Call-ID: 06507ce0125a89d472c143ef289473f1@192.168.0.133:5060
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:Айпи провайдера:5062 --->
CANCEL sip:38Номер на который звонили@Наш внешний айпи;user=phone SIP/2.0
Via: SIP/2.0/UDP Айпи провайдера:5062;rport;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 CANCEL
Max-Forwards: 70
User-Agent: TS-v4.5.1-16aW
Reason: SIP;cause=487;text="Request Terminated"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to Айпи провайдера:5062 (no NAT)

<--- Reliably Transmitting (no NAT) to Айпи провайдера:5062 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP Айпи провайдера:5062;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117;received=Айпи провайдера;rport=5062
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as7a68090b
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to Айпи провайдера:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP Айпи провайдера:5062;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117;received=Айпи провайдера;rport=5062
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as7a68090b
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 CANCEL
Server: FPBX-2.11.0(11.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.134:5160:
CANCEL sip:1038Номер на который звонили@192.168.0.134:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
Max-Forwards: 70
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060' in 6400 ms (Method: INVITE)
  == Spawn extension (ext-trunk, tdial, 9) exited non-zero on 'SIP/Provider-1-00000036'

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 CANCEL
User-Agent: dble
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.134:5160 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1073364917
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 INVITE
User-Agent: dble
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.134:5160:
ACK sip:9000@192.168.0.134:5160 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.133:5060;branch=z9hG4bK2bc4a4d4
Max-Forwards: 70
From: " Номер с которого звонили" <sip: Номер с которого звонили@192.168.0.133>;tag=as655599d7
To: <sip:1038Номер на который звонили@192.168.0.134:5160>;tag=1073364917
Contact: <sip: Номер с которого звонили@192.168.0.133:5060>
Call-ID: 7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.13.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:Айпи провайдера:5062 --->
ACK sip:38Номер на который звонили@Наш внешний айпи;user=phone SIP/2.0
Via: SIP/2.0/UDP Айпи провайдера:5062;rport;branch=z9hG4bK-3871489178-3843154708-1496094849-2097126117
From: <sip: Номер с которого звонили@Айпи провайдера:5062;user=phone>;tag=3500883098-3843154708-1496094849-2097126117
To: <sip:38Номер на который звонили@Наш внешний айпи;user=phone>;tag=as7a68090b
Call-ID: 9a3cabe414e311e581982c59e59aff7c@Айпи провайдера
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: TS-v4.5.1-16aW
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '9a3cabe414e311e581982c59e59aff7c@Айпи провайдера' Method: ACK
Really destroying SIP dialog '7e2f333c0bc003a2557367a40cb0dcc7@192.168.0.133:5060' Method: INVITE
Really destroying SIP dialog '06507ce0125a89d472c143ef289473f1@192.168.0.133:5060' Method: NOTIFY

<--- SIP read from UDP:192.168.0.134:5160 --->

<------------->

<--- SIP read from UDP:192.168.0.132:59580 --->

Причем первый вызов поймался 503 а второй нормально зашел.
Ну и старая проблема. Гудки в телефоне есть, а со шлюза на телефон вызов не падает.

Sparkle · Сообщение **Sparkle** » 19 июн 2015, 16:23

никто с таким не сталкивался?
Хотя бы в какую сторону копать, можете подсказать?(

Vlad1983 · Сообщение **Vlad1983** » 19 июн 2015, 18:14

Код: Выделить всё

    -- Executing [tdial@ext-trunk:9] Dial("SIP/Provider-2-00000052", "SIP/9000/10номер на который звонят,300,") in new stack

префикс "10" точно в шлюзе отрезается?
DIALING:тут только номер без всяких левых префиксов

Sparkle · Сообщение **Sparkle** » 21 июн 2015, 15:14

Да, в статусе диалинг номер без префикса, отрезается корректно.

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