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GSM Шлюз OpenVox VS-GW1600

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

Dober
Сообщения: 4
Зарегистрирован: 05 авг 2013, 10:48

GSM Шлюз OpenVox VS-GW1600

Сообщение Dober »

Всем хорошего настроения!
Скажите, сталкивался ли кто-то с таким зверем?
Интересует стабильность работы, ну, например, по сравнению с Portech
klistrod
Сообщения: 12
Зарегистрирован: 27 сен 2011, 20:57

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение klistrod »

Работает лучше чем Портеч и GoIP, очень хорошая тех поддержка от Китая, а настройки тут:
http://skytel.by/manual/openvox/119-nas ... -vs-gw1600
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

Приветствую всех!
купили данный девайс, теперь встал вопрос о настройки с GSM шлюза с FreePBX.
знаю выше есть инструкция, но все е хотел поинтересоваться, как будет "правльнее" настроить данную железку?
ведь она может выступать и в роли сервера и клиента и т.п. так как жлюз сам работает на астериске...
Аватара пользователя
SolarW
Сообщения: 1331
Зарегистрирован: 01 сен 2010, 14:21
Откуда: Днепропетровск, Украина

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение SolarW »

Из руководства приведенного по ссылке:
У каждой платы-модуля свой IP адрес:
Однако оригинальная схемотехника.
5 самостоятельных 4-х портовых GSM-шлюза каким-то внутренним свичем собранные в единую сеть?
Оригинально, оригинально...
Оригинальней этого было только у портеча - насколько помню отдельные GSM-шлюзы входящие в состав устройства наружу через NAT шли или еще какой-то подобный маразм был...
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

Ребят а как правильно настроить исходящую маршрутизацию?
Изображение
а то все вызовы идут через вторую симку...
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

но если в настройках шлюза отключить транк от второй симки и потом повторно включить то уже все вызовы исходящие ходят по первой симке...
если данный трюк повторить и первой симкой, то вызовы ходят по второй симке...
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

вот лог шлюза

Код: Выделить всё

[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Auto destroying SIP dialog '0e26369471e7a9b97d826c4658340e0a@0.0.0.0'
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Destroying SIP dialog 0e26369471e7a9b97d826c4658340e0a@0.0.0.0
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Auto destroying SIP dialog '22717c971993a04b6e7d6ddb69731168@0.0.0.0'
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Destroying SIP dialog 22717c971993a04b6e7d6ddb69731168@0.0.0.0
[Jun  6 11:50:18] DEBUG[1815] acl.c: For destination '192.168.100.2', our source address is '192.168.100.3'.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.100.3:5060
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Allocating new SIP dialog for 5d8246c67a03cb8900237e600fa47d92@192.168.100.2:5060 - INVITE (No RTP)
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Jun  6 11:50:18] DEBUG[1815] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer"
[Jun  6 11:50:18] DEBUG[1815] sip/reqresp_parser.c: Found SIP option: -replaces-
[Jun  6 11:50:18] DEBUG[1815] sip/reqresp_parser.c: Matched SIP option: replaces
[Jun  6 11:50:18] DEBUG[1815] sip/reqresp_parser.c: Found SIP option: -timer-
[Jun  6 11:50:18] DEBUG[1815] sip/reqresp_parser.c: Matched SIP option: timer
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: Splitting '192.168.100.2:5060' into...
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: ...host '192.168.100.2' and port '5060'.
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: Splitting '192.168.100.2' into...
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: ...host '192.168.100.2' and port ''.
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x2625f0'
[Jun  6 11:50:18] DEBUG[1815] res_rtp_asterisk.c: Allocated port 11352 for RTP instance '0x2625f0'
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: RTP instance '0x2625f0' is setup and ready to go
[Jun  6 11:50:18] DEBUG[1815] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x2625f0'
[Jun  6 11:50:18] VERBOSE[1815] netsock2.c:   == Using SIP RTP CoS mark 5
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Setting NAT on RTP to On
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing session-level SDP o=root 1041659241 1041659241 IN IP4 192.168.100.2... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing session-level SDP s=Asterisk PBX 11.7.0... UNSUPPORTED OR FAILED.
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: Splitting '192.168.100.2' into...
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: ...host '192.168.100.2' and port ''.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.100.2... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Setting payload 0 based on m type on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Setting payload 8 based on m type on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Setting payload 3 based on m type on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Setting payload 101 based on m type on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Incorporating payload 0 on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Incorporating payload 3 on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Incorporating payload 8 on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Incorporating payload 101 on 0xbcffa3bc
[Jun  6 11:50:18] DEBUG[1815] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2625f0'
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Copying payload 0 from 0xbcffa3bc to 0x26279c
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Copying payload 3 from 0xbcffa3bc to 0x26279c
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Copying payload 8 from 0xbcffa3bc to 0x26279c
[Jun  6 11:50:18] DEBUG[1815] rtp_engine.c: Copying payload 101 from 0xbcffa3bc to 0x26279c
[Jun  6 11:50:18] DEBUG[1815] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x2625f0'
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw)
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Checking SIP call limits for device meg_sim1
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Updating call counter for incoming call
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: Splitting '192.168.100.3' into...
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: ...host '192.168.100.3' and port ''.
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: Splitting '192.168.100.2' into...
[Jun  6 11:50:18] DEBUG[1815] netsock2.c: ...host '192.168.100.2' and port ''.
[Jun  6 11:50:18] DEBUG[1815] frame.c: Could not find preferred codec - Going for the best codec
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: *** Our native formats are 0x4 (ulaw) 
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) 
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: *** Our capabilities are 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) 
[Jun  6 11:50:18] DEBUG[1815] frame.c: Could not find preferred codec - Going for the best codec
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) 
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: This channel will not be able to handle video.
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: build_route: Contact hop: <sip:beeline_sim2@192.168.100.2:5060>
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: SIP/meg_sim1-192.168.100.2-00000017: New call is still down.... Trying... 
[Jun  6 11:50:18] DEBUG[1815] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.100.2:5060
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'NoOp'
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: No provider found, checking channel drivers for SIP - meg_sim1-192.168.100.2
[Jun  6 11:50:18] DEBUG[1767] chan_sip.c: Checking device state for peer meg_sim1-192.168.100.2
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: Changing state for SIP/meg_sim1-192.168.100.2 - state 1 (Not in use)
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: device 'SIP/meg_sim1-192.168.100.2' state '1'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sipinbound:1] NoOp("SIP/meg_sim1-192.168.100.2-00000017", "SIP Inbound") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'SIPROUTE' is 'sip-meg_sim1-192.168.100.2'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'EXTEN' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is '1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sipinbound:2] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "1?:nocdr") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Not taking any branch
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'SIPROUTE' is 'sip-meg_sim1-192.168.100.2'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'EXTEN' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Goto'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sipinbound:3] Goto("SIP/meg_sim1-192.168.100.2-00000017", "sip-meg_sim1-192.168.100.2,+79031234567,1") in new stack
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Goto (sip-meg_sim1-192.168.100.2,+79031234567,1)
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'NoOp'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sip-meg_sim1-192.168.100.2:1] NoOp("SIP/meg_sim1-192.168.100.2-00000017", "_[*#+0-9]. matches Rule rtg-meg_sim1-1") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'EXTEN' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sip-meg_sim1-192.168.100.2:2] Set("SIP/meg_sim1-192.168.100.2-00000017", "CDR_CALLEEID=+79031234567") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'EXTEN' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Macro'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [+79031234567@sip-meg_sim1-192.168.100.2:3] Macro("SIP/meg_sim1-192.168.100.2-00000017", "dial-failover,,+79031234567,extra/1,0,gsm-1") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:1] Set("SIP/meg_sim1-192.168.100.2-00000017", "ADEV=3") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:2] Set("SIP/meg_sim1-192.168.100.2-00000017", "AEXTEN_FLAG=4") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:3] Set("SIP/meg_sim1-192.168.100.2-00000017", "ACDR_NAME=5") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:4] Set("SIP/meg_sim1-192.168.100.2-00000017", "ARG=ARG") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:5] Set("SIP/meg_sim1-192.168.100.2-00000017", "MAX=128") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is 'ulaw'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'SIPAddHeader'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:6] SIPAddHeader("SIP/meg_sim1-192.168.100.2-00000017", "X-Best-Codec: ulaw") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: SIPAddHeader
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG1' is ''
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Expression result is '1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:7] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "1?dialstrnoforward") in new stack
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Goto (macro-dial-failover,s,14)
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Gotoif
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG2' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Expression result is '0'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:14] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "0?dialstrnoexten") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Not taking any branch
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Gotoif
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG2' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Expression result is '0'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:15] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "0?dialstrnoexten") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Not taking any branch
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Gotoif
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG' is 'ARG'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'AEXTEN_FLAG' is '4'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is 'ARG4'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG4' is '0'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Expression result is '0'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:16] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "0?dialstrnoexten") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Not taking any branch
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Gotoif
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG' is 'ARG'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ADEV' is '3'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is 'ARG3'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG3' is 'extra/1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG2' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:17] Set("SIP/meg_sim1-192.168.100.2-00000017", "DIALSTR=extra/1/+79031234567") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG' is 'ARG'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ADEV' is '3'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is 'ARG3'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG3' is 'extra/1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:18] Set("SIP/meg_sim1-192.168.100.2-00000017", "OUTDEV=extra/1") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'OUTDEV' is 'extra/1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Expression result is '0'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'GotoIf'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:19] GotoIf("SIP/meg_sim1-192.168.100.2-00000017", "0?exit,1") in new stack
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Not taking any branch
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Gotoif
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG' is 'ARG'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ACDR_NAME' is '5'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Function result is 'ARG5'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG5' is 'gsm-1'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:20] Set("SIP/meg_sim1-192.168.100.2-00000017", "CDR_TOCHAN=gsm-1") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'ARG2' is '+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Set'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:21] Set("SIP/meg_sim1-192.168.100.2-00000017", "CDR_CALLEEID=+79031234567") in new stack
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Set
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Goto'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:22] Goto("SIP/meg_sim1-192.168.100.2-00000017", "dial") in new stack
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Goto (macro-dial-failover,s,30)
[Jun  6 11:50:18] DEBUG[4117] app_macro.c: Executed application: Goto
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Result of 'DIALSTR' is 'extra/1/+79031234567'
[Jun  6 11:50:18] DEBUG[4117] pbx.c: Launching 'Dial'
[Jun  6 11:50:18] VERBOSE[4117] pbx.c:     -- Executing [s@macro-dial-failover:30] Dial("SIP/meg_sim1-192.168.100.2-00000017", "extra/1/+79031234567") in new stack
[Jun  6 11:50:18] DEBUG[4117] chan_extra.c: Using channel 1
[Jun  6 11:50:18] DEBUG[4117] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Jun  6 11:50:18] DEBUG[4117] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Jun  6 11:50:18] DEBUG[4117] devicestate.c: device 'EXTRA/1-1' state '2'
[Jun  6 11:50:18] DEBUG[4117] rtp_engine.c: Can't find native functions for channel 'EXTRA/1-1'
[Jun  6 11:50:18] DEBUG[4117] rtp_engine.c: Seeded SDP of 'EXTRA/1-1' with that of 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable DIALEDTIME.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ANSWEREDTIME.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable DIALEDPEERNAME.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable DIALEDPEERNUMBER.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable DIALSTATUS.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable MACRO_DEPTH.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable CDR_CALLEEID.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable CDR_TOCHAN.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable OUTDEV.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable DIALSTR.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Copying hard-transferable variable SIPADDHEADER01.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable MAX.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ACDR_NAME.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable AEXTEN_FLAG.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ADEV.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG5.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG4.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG3.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG2.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable ARG1.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable MACRO_PRIORITY.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable MACRO_CONTEXT.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable MACRO_EXTEN.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable SIPROUTE.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable SIPCALLID.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable SIPDOMAIN.
[Jun  6 11:50:18] DEBUG[4117] channel.c: Not copying variable SIPURI.
[Jun  6 11:50:18] VERBOSE[4117] chan_extra.c:     -- Requested transfer capability: 0x00 - SPEECH
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: No provider found, checking channel drivers for EXTRA - 1
[Jun  6 11:50:18] VERBOSE[4117] app_dial.c:     -- Called extra/1/+79031234567
[Jun  6 11:50:18] DEBUG[4117] channel.c: Set channel EXTRA/1-1 to read format ulaw
[Jun  6 11:50:18] DEBUG[4117] channel.c: Set channel SIP/meg_sim1-192.168.100.2-00000017 to read format alaw
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: Changing state for EXTRA/1 - state 2 (In use)
[Jun  6 11:50:18] DEBUG[1767] devicestate.c: device 'EXTRA/1' state '2'
[Jun  6 11:50:18] DEBUG[4099] chan_extra.c: Queuing frame from GSM_EVENT_PROCEEDING on channel 1 span 1
[Jun  6 11:50:18] VERBOSE[4117] app_dial.c:     -- EXTRA/1-1 is proceeding passing it to SIP/meg_sim1-192.168.100.2-00000017
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.100.2:5060
[Jun  6 11:50:18] VERBOSE[4117] app_dial.c:     -- EXTRA/1-1 is making progress passing it to SIP/meg_sim1-192.168.100.2-00000017
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: Setting framing from config on incoming call
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: ** Our prefcodec: 0x0 (nothing) 
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: -- Done with adding codecs to SDP
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw)
[Jun  6 11:50:18] DEBUG[4117] chan_sip.c: Trying to put 'SIP/2.0 183' onto UDP socket destined for 192.168.100.2:5060
[Jun  6 11:50:18] VERBOSE[4099] chan_extra.c:     -- PROGRESS with cause code 0 received
[Jun  6 11:50:18] DEBUG[4099] chan_extra.c: Queuing frame from GSM_EVENT_PROGRESS on channel 1 span 1
[Jun  6 11:50:18] VERBOSE[4117] app_dial.c:     -- EXTRA/1-1 is making progress passing it to SIP/meg_sim1-192.168.100.2-00000017
[Jun  6 11:50:18] VERBOSE[4117] app_dial.c:     -- EXTRA/1-1 is making progress passing it to SIP/meg_sim1-192.168.100.2-00000017
[Jun  6 11:50:18] DEBUG[4117] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Jun  6 11:50:18] DEBUG[4117] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Jun  6 11:50:18] DEBUG[4117] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x2625f0'
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: **** Received CANCEL (14) - Command in SIP CANCEL
[Jun  6 11:50:20] DEBUG[1815] netsock2.c: Splitting '192.168.100.2:5060' into...
[Jun  6 11:50:20] DEBUG[1815] netsock2.c: ...host '192.168.100.2' and port '5060'.
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: Setting SIP_ALREADYGONE on dialog 5d8246c67a03cb8900237e600fa47d92@192.168.100.2:5060
[Jun  6 11:50:20] DEBUG[1815] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2625f0'
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: Trying to put 'SIP/2.0 487' onto UDP socket destined for 192.168.100.2:5060
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.100.2:5060
[Jun  6 11:50:20] DEBUG[4117] channel.c: Deadlock avoided for write to channel 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jun  6 11:50:20] DEBUG[1815] chan_sip.c: Stopping retransmission on '5d8246c67a03cb8900237e600fa47d92@192.168.100.2:5060' of Response 102: Match Found
[Jun  6 11:50:20] DEBUG[4117] channel.c: Hanging up channel 'EXTRA/1-1'
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: extra_hangup(EXTRA/1-1)
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: Set option AUDIO MODE, value: ON(1) on EXTRA/1-1
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: Hangup: channel: 1 index = 0, normal = 10, callwait = -1, thirdcall = -1
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: Not yet hungup...  Calling hangup once with icause, and clearing call
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: Set option TDD MODE, value: OFF(0) on EXTRA/1-1
[Jun  6 11:50:20] DEBUG[4117] chan_extra.c: Set option AUDIO MODE, value: OFF(0) on EXTRA/1-1
[Jun  6 11:50:20] VERBOSE[4117] chan_extra.c:     -- Hungup 'EXTRA/1-1'
[Jun  6 11:50:20] DEBUG[4117] app_dial.c: Exiting with DIALSTATUS=CANCEL.
[Jun  6 11:50:20] DEBUG[4117] app_macro.c: Spawn extension (macro-dial-failover,s,30) exited non-zero on 'SIP/meg_sim1-192.168.100.2-00000017' in macro 'dial-failover'
[Jun  6 11:50:20] VERBOSE[4117] app_macro.c:   == Spawn extension (macro-dial-failover, s, 30) exited non-zero on 'SIP/meg_sim1-192.168.100.2-00000017' in macro 'dial-failover'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Spawn extension (sip-meg_sim1-192.168.100.2,+79031234567,3) exited non-zero on 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:20] VERBOSE[4117] pbx.c:   == Spawn extension (sip-meg_sim1-192.168.100.2, +79031234567, 3) exited non-zero on 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:20] DEBUG[4117] channel.c: Soft-Hanging up channel 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Function result is 'beeline_sim2'
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: No provider found, checking channel drivers for EXTRA - 1
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Result of 'CDR_CALLEEID' is '+79031234567'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Result of 'CDR_TOCHAN' is 'gsm-1'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Function result is '2014-06-06 11:50:18'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Function result is '0'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Function result is 'NO ANSWER'
[Jun  6 11:50:20] DEBUG[4117] pbx.c: Launching 'WriteCDR'
[Jun  6 11:50:20] VERBOSE[4117] pbx.c:     -- Executing [h@sip-meg_sim1-192.168.100.2:1] WriteCDR("SIP/meg_sim1-192.168.100.2-00000017", ""beeline_sim2","+79031234567","meg1234567","gsm-1","2014-06-06 11:50:18","0","NO ANSWER"") in new stack
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: Changing state for EXTRA/1 - state 0 (Unknown)
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: device 'EXTRA/1' state '0'
[Jun  6 11:50:20] DEBUG[4117] channel.c: Hanging up channel 'SIP/meg_sim1-192.168.100.2-00000017'
[Jun  6 11:50:20] DEBUG[4117] chan_sip.c: Hangup call SIP/meg_sim1-192.168.100.2-00000017, SIP callid 5d8246c67a03cb8900237e600fa47d92@192.168.100.2:5060
[Jun  6 11:50:20] DEBUG[4117] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x2625f0'
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: No provider found, checking channel drivers for SIP - meg_sim1-192.168.100.2
[Jun  6 11:50:20] DEBUG[1767] chan_sip.c: Checking device state for peer meg_sim1-192.168.100.2
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: Changing state for SIP/meg_sim1-192.168.100.2 - state 1 (Not in use)
[Jun  6 11:50:20] DEBUG[1767] devicestate.c: device 'SIP/meg_sim1-192.168.100.2' state '1'
[Jun  6 11:50:21] VERBOSE[4099] chan_extra.c:     -- Channel 1, span 1 received AOC-E charging 0 units
[Jun  6 11:50:21] DEBUG[1815] chan_sip.c: Destroying SIP dialog 5d8246c67a03cb8900237e600fa47d92@192.168.100.2:5060
[Jun  6 11:50:21] DEBUG[1815] rtp_engine.c: Destroyed RTP instance '0x2625f0'
[Jun  6 11:50:21] DEBUG[1815] chan_sip.c: Auto destroying SIP dialog '4187a84362937c3c1b0ab4ac14ece2ed@0.0.0.0'
[Jun  6 11:50:21] DEBUG[1815] chan_sip.c: Destroying SIP dialog 4187a84362937c3c1b0ab4ac14ece2ed@0.0.0.0
[Jun  6 11:50:21] DEBUG[1815] chan_sip.c: Auto destroying SIP dialog '0e4906e74722c5394ddbe13a3d117c48@0.0.0.0'
[Jun  6 11:50:21] DEBUG[1815] chan_sip.c: Destroying SIP dialog 0e4906e74722c5394ddbe13a3d117c48@0.0.0.0
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

Ребят я как понял у шлюза своя внутренняя очередь...
мб править родные файлы шлюза (он на астериске)
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

Код: Выделить всё

queues.conf
persistentmembers = yes
мб отключить?
a25k03
Сообщения: 93
Зарегистрирован: 20 сен 2013, 14:42

Re: GSM Шлюз OpenVox VS-GW1600

Сообщение a25k03 »

Возникла следующая проблема:
sim карта от мегафона не может зарегистрироваться в сети, пробовал в 4 слотах...
НО! если взять и вставить симку в телефон, то в нем симка норм работает.
Далее выключаю телефон, переставляю sim карту в шлюз и там он регистрируется на минуту-две и далее теряет сеть и опять не может зарегистрироваться....
пробовал менять imei, 8, 12 и последнюю цифру - безрезультатно...
оператору звонил безрезультатно...
единственно известно то что после грозы началась данная проблема.
и все абоненты мегафона в данной части города жалуются на качество связи...

другие симки работают исправно на данном шлюзе...

вопрос: это заблокировал оператор данный шлюз или глюк их оборудования?
Ответить
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