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ReloadXP
СообщениеДобавлено: Пн сен 28, 2015 15:39 
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Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
Добрый день
Возникла проблема с DVG 6008s
DVG 6008s - не отключается от линии при исходящем звонке!
Когда слышит от абонента сигнал занято, то ложит трубку. А так постоянно статус Talking VOIP OUT и дозвон до клиента.
Т.е. от астериска шлюз не понимает или не принимает сигнал - отрубись...
Как исправить проблему? Или куда копать хоть? 2 дня уже мучаюсь.


Телефония Asterisk 11.7.0~dfsg-1ubuntu1
Линия цифровая с поддержкой CallerID
Сип-клиент Bria 3.5.5
Маршрутизатор Microtik -

Что сделано:
1. Проброшены порты до сервера Астериск 5060, 5061, 10000-20000, астериск настроен.
2. Присвоены не изменяющиеся IP адреса шлюзу и астериск
3. Правило набора в Астериск
Скрытый текст: показать
[dlink_outcoming]
exten => _X!,1(chan1011),Dial(SIP/1011/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1012)
exten => _X!,n,Hangup

exten => _X!,n(chan1012),Dial(SIP/1012/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1013)
exten => _X!,n,Hangup

exten => _X!,n(chan1013),Dial(SIP/1013/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1014)
exten => _X!,n,Hangup

exten => _X!,n(chan1014),Dial(SIP/1014/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1015)
exten => _X!,n,Hangup

exten => _X!,n(chan1015),Dial(SIP/1015/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1016)
exten => _X!,n,Hangup

exten => _X!,n(chan1016),Dial(SIP/1016/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1017)
exten => _X!,n,Hangup

exten => _X!,n(chan1017),Dial(SIP/1017/${EXTEN})
exten => _X!,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?chan1018)
exten => _X!,n,Hangup

exten => _X!,n(chan1018),Dial(SIP/1018/${EXTEN})
exten => _X!,n,Hangup

4. Оттестированы входящие и исходящие звонки. Номера определются, всё настроил.

Проблема следующая.


== Using SIP RTP CoS mark 5
-- Executing [1234567890@operator:1] Goto("SIP/2000-00000008", "dlink_outcoming,1234567890,1") in new stack
-- Goto (dlink_outcoming,1234567890,1)
-- Executing [1234567890@dlink_outcoming:1] Dial("SIP/2000-00000008", "SIP/1011/1234567890") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1011/1234567890
-- SIP/1011-00000009 answered SIP/2000-00000008
-- Locally bridging SIP/2000-00000008 and SIP/1011-00000009
> 0x7fe82c014110 -- Probation passed - setting RTP source address to 192.168.1.11:9000 - это что за RTP? Как сделать чтобы оно не выходило из диапазона 10000-20000 ?
> 0x7fe82002f4c0 -- Probation passed - setting RTP source address to МойАЙПИ:51814

ЛОЖУ ТРУПКУ НА СИП
== Spawn extension (dlink_outcoming, 1234567890, 1) exited non-zero on 'SIP/2000-0000000a'
ЗВОНОК НЕ СБРАСЫВАЕТСЯ, Шлюз IP продолжает звонить! 1234567890 - мой номер.

DVG 6008s - не отключается от линии при исходящем звонке!


Вот скрины:


Вложения:
Комментарий к файлу: slmon log
456.png
456.png [ 48.1 KiB | Просмотров: 5921 ]
Скриншот 2015-09-28 15.50.44.png
Скриншот 2015-09-28 15.50.44.png [ 227.76 KiB | Просмотров: 5927 ]
Скриншот 2015-09-28 15.42.42.png
Скриншот 2015-09-28 15.42.42.png [ 215.83 KiB | Просмотров: 5927 ]


Последний раз редактировалось ReloadXP Пн сен 28, 2015 17:22, всего редактировалось 3 раз(а).
Вернуться наверх
 Профиль  
 
ReloadXP
СообщениеДобавлено: Пн сен 28, 2015 15:53 
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Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
И вот это может связано?

The last packet's source IP 192.168.1.18 The last packet's source Port 18365
Потеря пакетов с Астериском
Вернуться наверх
 Профиль  
 
ReloadXP
СообщениеДобавлено: Пн сен 28, 2015 17:25 
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Зарегистрирован: Пн сен 28, 2015 14:49
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Техподдержка существует тут?
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 Профиль  
 
Evgeniy Ryzhov
СообщениеДобавлено: Вт сен 29, 2015 12:04 
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Сотрудник D-LINK
Сотрудник D-LINK

Зарегистрирован: Ср янв 22, 2014 18:37
Сообщений: 1158
Здравствуйте
Какая верси прошивки на шлюзе? и нужен дамп трафика со шлюза.
Если дамп снять не получится, то лог slmon с командами: Advance -> Manual Command -> Command Code 999 и Parametr 1
Вернуться наверх
 Профиль  
 
ReloadXP
СообщениеДобавлено: Пт окт 02, 2015 17:29 
Не в сети

Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
Evgeniy Ryzhov писал(а):
Здравствуйте
Какая верси прошивки на шлюзе? и нужен дамп трафика со шлюза.
Если дамп снять не получится, то лог slmon с командами: Advance -> Manual Command -> Command Code 999 и Parametr 1



Current Software Version No. [1.02.38.95]

Логи с командами Command Code 999 и Parametr 1:
18:25:24 [011016] Receive Cmd=999, ParaLen=1, Para=[1]
18:25:30 [011082]
18:25:30 [011082] User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
18:25:30 [011082] Date: Fri, 02 Oct 2015 15:25:39 GMT
18:25:30 [011082] a=rtpmap:18 G729/8000
18:25:31 [011082]
18:25:31 [011083] Content-Type: application/sdp
18:25:31 [011083] 9: 1011=OFFERING
18:25:31 [011083] 9: Get CallerId=2000
18:25:31 [011083] 9: Hunting Trunk Line
18:25:31 [011083] 0: Peer=192.168.1.18#12720, PT=18, RecvOnly=0
18:25:31 [011083] 0: TrunkPrefix=, Dest=83852251906, Dialno=83852251906
18:25:31 [011083] 0: FxoHookOff
18:25:31 [011083] 0: InputGain(-2)
18:25:31 [011083] 0: ==18:TrunkDialOut
18:25:31 [011084]
18:25:31 [011084] <---- 192.168.1.18:5060
18:25:31 [011084] INVITE sip:83852251906@192.168.1.11 SIP/2.0
18:25:31 [011084] Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK45a823b5;rport
18:25:31 [011084] Max-Forwards: 70
18:25:31 [011084] From: "2000" <sip:2000@192.168.1.18>;tag=as7e6d9aec
18:25:31 [011084] To: <sip:83852251906@192.168.1.11>
18:25:31 [011084] Contact: <sip:2000@192.168.1.18:5060>
18:25:31 [011084] Call-ID: 31c492d014853a185e9ef5ad19e7e69d@192.168.1.18:5060
18:25:31 [011084] CSeq: 102 INVITE
18:25:31 [011084] User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
18:25:31 [011084] Date: Fri, 02 Oct 2015 15:25:39 GMT
18:25:31 [011084] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
18:25:31 [011084] Supported: replaces, timer
18:25:31 [011084] Content-Type: application/sdp
18:25:31 [011084] Content-Length: 341
18:25:31 [011084] v=0
18:25:31 [011084] o=root 678091587 678091587 IN IP4 192.168.1.18
18:25:31 [011084] s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
18:25:31 [011084] c=IN IP4 192.168.1.18
18:25:31 [011084] t=0 0
18:25:31 [011084] m=audio 12720 RTP/AVP 18 3 0 8 101
18:25:31 [011084] a=rtpmap:18 G729/8000
18:25:31 [011084] a=fmtp:18 annexb=no
18:25:31 [011084] a=rtpmap:3 GSM/8000
18:25:31 [011084] a=rtpmap:0 PCMU/8000
18:25:31 [011084] a=rtpmap:8 PCMA/8000
18:25:31 [011084] a=rtpmap:101 telephone-event/8000
18:25:31 [011084] a=fmtp:101 0-16
18:25:31 [011084] a=ptime:20
18:25:31 [011084] a=sendrecv
18:25:31 [011084] -----
18:25:31 [011084]
18:25:31 [011084]
18:25:31 [011084] ----> 192.168.1.18:5060
18:25:31 [011084] SIP/2.0 100 Trying
18:25:31 [011084] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK45a823b5
18:25:31 [011084] From: "2000" <sip:2000@192.168.1.18>;tag=as7e6d9aec
18:25:31 [011084] To: <sip:83852251906@192.168.1.11>
18:25:31 [011085] Call-ID: 31c492d014853a185e9ef5ad19e7e69d@192.168.1.18:5060
18:25:31 [011085] CSeq: 102 INVITE
18:25:31 [011085] Content-Type: application/sdp
18:25:31 [011085] Content-Length: 0
18:25:31 [011085] -----
18:25:31 [011085]
18:25:32 [011094] 0: StopPlayTone()=0
18:25:32 [011094] 0: DialOut(83852251906)=0
18:25:34 [011114] 9: 1011=ACCEPT
18:25:34 [011114] 0: ==13:VoipAnswering
18:25:34 [011115]
18:25:34 [011115] ----> 192.168.1.18:5060
18:25:34 [011115] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK45a823b5
18:25:34 [011115] s=Session SDP
18:25:34 [011116] 9: 1011=CONNECTED
18:25:34 [011116] 0: StopDeviceVoiceTone
18:25:34 [011116] 0: SetVoiceCoder(17)=0
18:25:34 [011116] 0: localip = 192.168.1.11 !!
18:25:34 [011116] 0: RTP[1,1], Peer=192.168.1.18#12720, PT=18/2/0, local=192.168.1.11#0
18:25:34 [011116] 0: SetTalkMode[1,1]
18:25:34 [011116] 0: ==14:Talking
18:25:50 [011280]
18:25:50 [011280] <---- 192.168.1.18:5060
18:25:50 [011280] BYE sip:1011@192.168.1.11:5060 SIP/2.0
18:25:50 [011280] Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK21597f57;rport
18:25:50 [011280] Max-Forwards: 70
18:25:50 [011280] From: "2000" <sip:2000@192.168.1.18>;tag=as7e6d9aec
18:25:50 [011280] To: <sip:83852251906@192.168.1.11>;tag=656da054-685920
18:25:50 [011280] Call-ID: 31c492d014853a185e9ef5ad19e7e69d@192.168.1.18:5060
18:25:50 [011280] CSeq: 103 BYE
18:25:50 [011280] User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
18:25:50 [011280] X-Asterisk-HangupCause: Normal Clearing
18:25:50 [011280] X-Asterisk-HangupCauseCode: 16
18:25:50 [011280] Content-Length: 0
18:25:50 [011280] -----
18:25:50 [011280]
18:25:50 [011281]
18:25:50 [011281] ----> 192.168.1.18:5060
18:25:50 [011281] SIP/2.0 481 Call Leg/Transaction Does Not Exist
18:25:50 [011281] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK21597f57
18:25:50 [011281] From: "2000" <sip:2000@192.168.1.18>;tag=as7e6d9aec
18:25:50 [011281] To: <sip:83852251906@192.168.1.11>;tag=656da054-685920
18:25:50 [011281] Call-ID: 31c492d014853a185e9ef5ad19e7e69d@192.168.1.18:5060
18:25:50 [011281] CSeq: 103 BYE
18:25:51 [011281] User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
18:25:51 [011281] Content-Length: 0
18:25:51 [011281] -----
18:25:51 [011281]
18:26:02 [011399] ----> 192.168.1.18:5060
18:26:02 [011399] -----
18:26:02 [011399] 0: SetFax(1)=0
18:26:05 [011430]
18:26:05 [011430] <---- 192.168.1.18:5060
18:26:05 [011430] OPTIONS sip:1011@192.168.1.11:5060 SIP/2.0
18:26:05 [011430] Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3c09aad9;rport
18:26:05 [011430] Max-Forwards: 70
18:26:05 [011431] From: "asterisk" <sip:asterisk@192.168.1.18>;tag=as15be26a5
18:26:05 [011431] To: <sip:1011@192.168.1.11:5060>
18:26:05 [011431] Contact: <sip:asterisk@192.168.1.18:5060>
18:26:05 [011431] Call-ID: 03ca265c4b4ad23567ea3ac0436feeb0@192.168.1.18:5060
18:26:05 [011431] CSeq: 102 OPTIONS
18:26:05 [011431] User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
18:26:05 [011431] Date: Fri, 02 Oct 2015 15:26:14 GMT
18:26:05 [011431] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
18:26:05 [011431] Supported: replaces, timer
18:26:05 [011431] Content-Length: 0
18:26:05 [011431] -----
18:26:05 [011431]
18:26:05 [011431]
18:26:05 [011431] ----> 192.168.1.18:5060
18:26:06 [011431] SIP/2.0 200 OK
18:26:06 [011431] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK3c09aad9
18:26:06 [011431] From: "asterisk" <sip:asterisk@192.168.1.18>;tag=as15be26a5
18:26:06 [011431] To: <sip:1011@192.168.1.11:5060>;tag=7d54cf61-685951
18:26:06 [011431] Call-ID: 03ca265c4b4ad23567ea3ac0436feeb0@192.168.1.18:5060
18:26:06 [011431] CSeq: 102 OPTIONS
18:26:06 [011431] Contact: <sip:1011@192.168.1.11:5060>
18:26:06 [011431] User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
18:26:06 [011431] Content-Length: 0
18:26:06 [011431] -----
18:26:06 [011431]

А трубка не высвобождается. Висела пока я не послал в линию шлюза "BUSY".


А теперь ещё интереснее. Когда звонок обратный, т.е. НА ШЛЮЗ и СИП КЛИЕНТ вешает трубку - BYE. Звонок успешно завершается.

Скрытый текст: показать
19:12:18 [039153]
19:12:18 [039153] ----> 192.168.1.18:5060
19:12:18 [039153] SIP/2.0 200 OK
19:12:18 [039153] Content-Length: 0
19:12:50 [039479] 0: Fxo Ringing
19:12:50 [039480] 0: RingTime(1040) First Stop
19:12:52 [039495] 0: RingTime(2600) Second Ring
19:12:54 [039517] 0: StopPlayTone()=0
19:12:54 [039517] 0: ### Fxo Get Fsk CallerId [3852380498,], Len=10
19:12:54 [039517] 0: CallerId=3852380498
19:12:54 [039517] 0: ==17:WaitAnswerDeviceOk
19:12:54 [039518] 0: StopDeviceVoiceTone
19:12:54 [039518] 0: HotLineEnabled with [1011]
19:12:54 [039518] 0: End Input Default Route [1]
19:12:54 [039518] 0: PhoneBook has GwNo=1011, addr=192.168.1.18:5060
19:12:54 [039518] 0: ==7:Inviting
19:12:54 [039518] 0: VoIP CallOut, Invite <sip:1011@192.168.1.18;user=phone>
19:12:54 [039519] 0: 1011=DIALING
19:12:54 [039520]
19:12:54 [039520] ----> 192.168.1.18:5060
19:12:54 [039520] INVITE sip:1011@192.168.1.18;user=phone SIP/2.0
19:12:54 [039520] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK4c108ea9ebaec5ee
19:12:54 [039520] From: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:12:54 [039520] To: <sip:1011@192.168.1.18;user=phone>
19:12:54 [039520] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:12:54 [039520] Session-Expires: 100
19:12:54 [039520]
19:12:54 [039520] <---- 192.168.1.18:5060
19:12:54 [039520] SIP/2.0 401 Unauthorized
19:12:54 [039520] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK4c108ea9ebaec5ee;received=192.168.1.11;rport=5060
19:12:54 [039520] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
19:12:54 [039522]
19:12:54 [039522] ----> 192.168.1.18:5060
19:12:54 [039522] ACK sip:1011@192.168.1.18;user=phone SIP/2.0
19:12:54 [039522] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK4c108ea9ebaec5ee
19:12:54 [039522] From: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:12:54 [039522] To: <sip:1011@192.168.1.18;user=phone>;tag=as11904a0c
19:12:54 [039522] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:12:54 [039522] CSeq: 16 INVITE
19:12:54 [039522] Content-Type: application/sdp
19:12:54 [039522] Content-Length: 331
19:12:54 [039522] v=0
19:12:54 [039522] o=1011 1795955420 1795955420 IN IP4 192.168.1.11
19:12:55 [039522] s=Session SDP
19:12:55 [039522] c=IN IP4 192.168.1.11
19:12:55 [039522] t=0 0
19:12:55 [039522] m=audio 11000 RTP/AVP 18 4 98 0 8
19:12:55 [039522] a=rtpmap:98 G726-32/8000
19:12:55 [039523]
19:12:55 [039523] <---- 192.168.1.18:5060
19:12:55 [039523] SIP/2.0 401 Unauthorized
19:12:55 [039523] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK4c108ea9ebaec5ee;received=192.168.1.11;rport=5060
19:12:55 [039523] From: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:12:55 [039523] To: <sip:1011@192.168.1.18;user=phone>;tag=as11904a0c
19:12:55 [039523] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:12:55 [039523] CSeq: 15 INVITE
19:12:55 [039523] Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
19:12:55 [039523] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
19:12:55 [039523] Supported: replaces, timer
19:12:55 [039523] WWW-Authenticate: Digest algorithm=MD5, realm="taxi2.asuscomm.com", nonce="2b8cc9a2"
19:12:55 [039523] Content-Length: 0
19:12:55 [039523] -----
19:12:55 [039523]
19:12:55 [039523]
19:12:55 [039523] ----> 192.168.1.18:5060
19:12:55 [039523] ACK sip:1011@192.168.1.18;user=phone SIP/2.0
19:12:55 [039523] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK4c108ea9ebaec5ee
19:12:55 [039524] SIP/2.0 100 Trying
19:12:55 [039524] Contact: <sip:1011@192.168.1.18:5060>
19:12:55 [039524] 0: 1011=PROCEEDING
19:13:10 [039674]
19:13:10 [039674] <---- 192.168.1.18:5060
19:13:10 [039674] SIP/2.0 200 OK
19:13:10 [039674] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bKa37a3924345d70ba;received=192.168.1.11;rport=5060
19:13:10 [039674] From: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:13:10 [039674] To: <sip:1011@192.168.1.18;user=phone>;tag=as4706fd59
19:13:10 [039674] Session-Expires: 100;refresher=uas
19:13:10 [039674] a=fmtp:18 annexb=no
19:13:10 [039676] 0: FxoHookOff
19:13:10 [039676]
19:13:10 [039676] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:13:10 [039677]
19:13:10 [039678]
19:13:10 [039678] ----> 192.168.1.18:5060
19:13:10 [039678] ACK sip:1011@192.168.1.18:5060 SIP/2.0
19:13:10 [039678] Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK1ad55a791db621e2
19:13:10 [039678] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:13:17 [039754]
19:13:17 [039754] <---- 192.168.1.18:5060
19:13:18 [039754] OPTIONS sip:1011@192.168.1.11:5060 SIP/2.0
19:13:18 [039754] Contact: <sip:asterisk@192.168.1.18:5060>
19:13:18 [039754]
19:13:18 [039754] ----> 192.168.1.18:5060
19:13:18 [039754] SIP/2.0 200 OK
19:13:18 [039754] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK7cd6845f
19:13:18 [039754] From: "asterisk" <sip:asterisk@192.168.1.18>;tag=as3e2372a2
19:13:18 [039754] To: <sip:1011@192.168.1.11:5060>;tag=ade2b841-688784
19:13:18 [039754] Call-ID: 3e9aa3de513ec83c4585deb05668a676@192.168.1.18:5060
19:13:18 [039754] CSeq: 102 OPTIONS
19:13:18 [039754] -----
19:13:18 [039755]
19:13:21 [039787]
19:13:21 [039787] <---- 192.168.1.18:5060
19:13:21 [039787] BYE sip:1011@192.168.1.11:5060 SIP/2.0
19:13:21 [039787] Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK7c65c124;rport
19:13:21 [039787] Max-Forwards: 70
19:13:21 [039787] From: <sip:1011@192.168.1.18;user=phone>;tag=as4706fd59
19:13:21 [039787] To: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:13:21 [039787] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:13:21 [039787] CSeq: 102 BYE
19:13:21 [039787] User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
19:13:21 [039787] Proxy-Authorization: Digest username="1011", realm="taxi2.asuscomm.com", algorithm=MD5, uri="sip:192
19:13:21 [039788] .168.1.18", nonce="2b8cc9a2", response="d07a258f24bee1861cf7d2adc0b3f108"
19:13:21 [039788] X-Asterisk-HangupCause: Unknown
19:13:21 [039788] X-Asterisk-HangupCauseCode: 0
19:13:21 [039788] Content-Length: 0
19:13:21 [039788] -----
19:13:21 [039788]
19:13:21 [039788] 0: 1011=DISCONNECT
19:13:21 [039788] 0: Rtp Talk Stop !
19:13:21 [039788] 0: ==15:PlayBusyTone
19:13:21 [039788]
19:13:21 [039789] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK7c65c124
19:13:21 [039789] 0: StopPlayTone()=0
19:13:21 [039789] 0: FxoHookOn
19:13:21 [039789] 0: SetFax(1)=0
19:13:21 [039789]
19:13:21 [039789] To: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:13:21 [039789]
19:13:21 [039790]
19:13:21 [039790] ----> 192.168.1.18:5060
19:13:21 [039790] SIP/2.0 200 OK
19:13:21 [039790] Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK7c65c124
19:13:21 [039790] From: <sip:1011@192.168.1.18;user=phone>;tag=as4706fd59
19:13:21 [039790] To: <sip:3852380498@192.168.1.18;user=phone>;tag=70fc2158-688760
19:13:21 [039790] Call-ID: D1B9-121C-4668876030B5C11A538A-002@SipHost
19:13:21 [039790] CSeq: 102 BYE
19:13:21 [039790] User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
19:13:21 [039790] Content-Length: 0
19:13:21 [039790] -----
19:13:21 [039790]
19:14:18 [040355] <---- 192.168.1.18:5060
19:14:18 [040355] Date: Fri, 02 Oct 2015 16:14:26 GMT
19:15:18 [040956]
19:15:18 [040956] <---- 192.168.1.18:5060
19:15:18 [040956] OPTIONS sip:1011@192.168.1.11:5060 SIP/2.0
19:15:18 [040956] Max-Forwards: 70
19:15:18 [040956] Content-Length: 0
19:15:18 [040956]
19:15:18 [040956] ----> 192.168.1.18:5060
19:15:18 [040956] SIP/2.0 200 OK
19:15:18 [040956] User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
Вернуться наверх
 Профиль  
 
ReloadXP
СообщениеДобавлено: Пн окт 05, 2015 08:16 
Не в сети

Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
///
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 Профиль  
 
Evgeniy Ryzhov
СообщениеДобавлено: Пн окт 05, 2015 12:27 
Не в сети
Сотрудник D-LINK
Сотрудник D-LINK

Зарегистрирован: Ср янв 22, 2014 18:37
Сообщений: 1158
Шлюз не принимает пакет BYE от астериска, потому что считает его не принадлежащим текущей сессии/транзакции или дублирующим. Нужен всё-таки дамп трафика с порта WAN шлюза.
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 Профиль  
 
ReloadXP
СообщениеДобавлено: Ср окт 07, 2015 12:54 
Не в сети

Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
Evgeniy Ryzhov писал(а):
Шлюз не принимает пакет BYE от астериска, потому что считает его не принадлежащим текущей сессии/транзакции или дублирующим. Нужен всё-таки дамп трафика с порта WAN шлюза.




!!!
Скрытый текст: показать
Лог с астериска

======== КОММЕНТАРИЙ - НАЧИНАЮ ЗВОНОК ======================================


<--- SIP read from UDP:10.8.0.7:13994 --->
INVITE sip:83852251906@10.8.1.1;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-2c8bba1aacf04b23-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.8.0.7:13994;transport=udp>
To: <sip:83852251906@10.8.1.1>
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.5 stamp 71238
Content-Length: 252

v=0
o=- 13088684725067418 1 IN IP4 10.8.0.7
s=Bria 3 release 3.5.5 stamp 71238
c=IN IP4 10.8.0.7
t=0 0
m=audio 14958 RTP/AVP 0 8 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 10.8.0.7:13994 (no NAT)
Sending to 10.8.0.7:13994 (no NAT)
Using INVITE request as basis request - MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
Found peer '2000' for '2000' from 10.8.0.7:13994

<--- Reliably Transmitting (NAT) to 10.8.0.7:13994 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-2c8bba1aacf04b23-1---d8754z-;received=10.8.0.7;rport=13994
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
To: <sip:83852251906@10.8.1.1>;tag=as79182c1e
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 1 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="lada.barnaul-it.com", nonce="10f28601"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.8.0.7:13994 --->
ACK sip:83852251906@10.8.1.1;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-2c8bba1aacf04b23-1---d8754z-;rport
Max-Forwards: 70
To: <sip:83852251906@10.8.1.1>;tag=as79182c1e
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.8.0.7:13994 --->
INVITE sip:83852251906@10.8.1.1;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-653eb6556cff537f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.8.0.7:13994;transport=udp>
To: <sip:83852251906@10.8.1.1>
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: Bria 3 release 3.5.5 stamp 71238
Authorization: Digest username="2000",realm="lada.barnaul-it.com",nonce="10f28601",uri="sip:83852251906@10.8.1.1;transport=udp",response="74f7b67b6048e43f2ec64d9900783243",algorithm=MD5
Content-Length: 252

v=0
o=- 13088684725067418 1 IN IP4 10.8.0.7
s=Bria 3 release 3.5.5 stamp 71238
c=IN IP4 10.8.0.7
t=0 0
m=audio 14958 RTP/AVP 0 8 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 10.8.0.7:13994 (NAT)
Using INVITE request as basis request - MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
Found peer '2000' for '2000' from 10.8.0.7:13994
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.8.0.7:14958
Looking for 83852251906 in operator (domain 10.8.1.1)
list_route: hop: <sip:2000@10.8.0.7:13994;transport=udp>

<--- Transmitting (NAT) to 10.8.0.7:13994 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-653eb6556cff537f-1---d8754z-;received=10.8.0.7;rport=13994
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
To: <sip:83852251906@10.8.1.1>
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 2 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:83852251906@10.8.0.101:5060>
Content-Length: 0


<------------>
-- Executing [83852251906@operator:1] Goto("SIP/2000-0000000e", "dlink_outcoming,83852251906,1") in new stack
-- Goto (dlink_outcoming,83852251906,1)
-- Executing [83852251906@dlink_outcoming:1] Dial("SIP/2000-0000000e", "SIP/1011/83852251906") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14428
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.11:5060:
INVITE sip:83852251906@192.168.1.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK644b4d23;rport
Max-Forwards: 70
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>
Contact: <sip:2000@192.168.1.18:5060>
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 07 Oct 2015 09:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 941553797 941553797 IN IP4 192.168.1.18
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.1.18
t=0 0
m=audio 14428 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/1011/83852251906
Retransmitting #1 (NAT) to 192.168.1.11:5060:
INVITE sip:83852251906@192.168.1.11 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK644b4d23;rport
Max-Forwards: 70
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>
Contact: <sip:2000@192.168.1.18:5060>
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 07 Oct 2015 09:45:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 941553797 941553797 IN IP4 192.168.1.18
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 192.168.1.18
t=0 0
m=audio 14428 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK644b4d23
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK644b4d23
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK644b4d23
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 INVITE
Contact: <sip:1011@192.168.1.11:5060>
User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
Content-Type: application/sdp
Content-Length: 237

v=0
o=1011 1799489540 1799489540 IN IP4 192.168.1.11
s=Session SDP
c=IN IP4 192.168.1.11
t=0 0
m=audio 9005 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.11:9005
list_route: hop: <sip:1011@192.168.1.11:5060>
set_destination: Parsing <sip:1011@192.168.1.11:5060> for address/port to send to
set_destination: set destination to 192.168.1.11:5060
Transmitting (NAT) to 192.168.1.11:5060:
ACK sip:1011@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK458957f8;rport
Max-Forwards: 70
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
Contact: <sip:2000@192.168.1.18:5060>
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0


---
-- SIP/1011-0000000f answered SIP/2000-0000000e
Audio is at 19440
Adding codec 100008 (g729) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 10.8.0.7:13994 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-653eb6556cff537f-1---d8754z-;received=10.8.0.7;rport=13994
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
To: <sip:83852251906@10.8.1.1>;tag=as1ba4c835
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 2 INVITE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:83852251906@10.8.0.101:5060>
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 1259204530 1259204530 IN IP4 10.8.0.101
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 10.8.0.101
t=0 0
m=audio 19440 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Locally bridging SIP/2000-0000000e and SIP/1011-0000000f

<--- SIP read from UDP:10.8.0.7:13994 --->
ACK sip:83852251906@10.8.0.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-d719fb4694f89540-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.8.0.7:13994;transport=udp>
To: <sip:83852251906@10.8.1.1>;tag=as1ba4c835
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 2 ACK
User-Agent: Bria 3 release 3.5.5 stamp 71238
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
> 0x7f04c4013400 -- Probation passed - setting RTP source address to 192.168.1.11:9005
> 0x7f04b8010af0 -- Probation passed - setting RTP source address to 10.8.0.7:14958

<--- SIP read from UDP:10.8.0.7:13994 --->


<------------->
Really destroying SIP dialog '110d29afabe4bff492123d902fb14a3b' Method: ACK


======== КОММЕНТАРИЙ - ПОЛОЖИЛ ТРУБКУ НА СИП КЛИЕНТЕ ======================================


<--- SIP read from UDP:10.8.0.7:13994 --->
BYE sip:83852251906@10.8.0.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-ee462d1e4c696d1a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.8.0.7:13994;transport=udp>
To: <sip:83852251906@10.8.1.1>;tag=as1ba4c835
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 3 BYE
User-Agent: Bria 3 release 3.5.5 stamp 71238
Authorization: Digest username="2000",realm="lada.barnaul-it.com",nonce="10f28601",uri="sip:83852251906@10.8.0.101:5060",response="367e28cfc9a28d11f2fb26d6f655c722",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.8.0.7:13994 (NAT)
Scheduling destruction of SIP dialog 'MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 10.8.0.7:13994 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.7:13994;branch=z9hG4bK-d8754z-ee462d1e4c696d1a-1---d8754z-;received=10.8.0.7;rport=13994
From: "2000"<sip:2000@10.8.1.1>;tag=4f55383c
To: <sip:83852251906@10.8.1.1>;tag=as1ba4c835
Call-ID: MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I
CSeq: 3 BYE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060' in 6656 ms (Method: INVITE)
set_destination: Parsing <sip:1011@192.168.1.11:5060> for address/port to send to
set_destination: set destination to 192.168.1.11:5060
Reliably Transmitting (NAT) to 192.168.1.11:5060:
BYE sip:1011@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK218865d3;rport
Max-Forwards: 70
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (dlink_outcoming, 83852251906, 1) exited non-zero on 'SIP/2000-0000000e'

<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK218865d3
From: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
To: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 103 BYE
User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060' Method: INVITE


======== КОММЕНТАРИЙ - СБРОСИЛ ВЫЗОВ - СИГНАЛ ЗАНЯТО В ЛИНИЮ =================================


Really destroying SIP dialog 'MDVlY2ZkMmMyOTc0ZDFkMmVjYzVkOWZkOWU5ZmI2N2I' Method: BYE

<--- SIP read from UDP:192.168.1.11:5060 --->
BYE sip:2000@192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK76c049d737ffee3c
From: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
To: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 14 BYE
Contact: <sip:1011@192.168.1.11:5060>
Max-Forwards:70
User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.11:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.11:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK76c049d737ffee3c;received=192.168.1.11
From: <sip:83852251906@192.168.1.11>;tag=d898d38e-692294
To: "2000" <sip:2000@192.168.1.18>;tag=as265633f7
Call-ID: 4dccdabc364733144d01cf0c04ca8177@192.168.1.18:5060
CSeq: 14 BYE
Server: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 192.168.1.11:5060:
OPTIONS sip:1011@192.168.1.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK559e9dea;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.18>;tag=as092d287b
To: <sip:1011@192.168.1.11:5060>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 31fb228558c37f63708480225fda157f@192.168.1.18:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 07 Oct 2015 09:46:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bK559e9dea
From: "asterisk" <sip:asterisk@192.168.1.18>;tag=as092d287b
To: <sip:1011@192.168.1.11:5060>;tag=e19d757d-692336
Call-ID: 31fb228558c37f63708480225fda157f@192.168.1.18:5060
CSeq: 102 OPTIONS
Contact: <sip:1011@192.168.1.11:5060>
User-Agent: dlink 12-3895-8558-1.2.1.2178-SA7O8
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '31fb228558c37f63708480225fda157f@192.168.1.18:5060' Method: OPTIONS

<--- SIP read from UDP:10.8.0.7:13994 --->


<------------->
Reliably Transmitting (NAT) to 10.8.0.7:13994:
OPTIONS sip:2000@10.8.0.7:13994;rinstance=7e938a4735992dc7;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.8.0.101:5060;branch=z9hG4bK28967712;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.8.0.101>;tag=as2586effb
To: <sip:2000@10.8.0.7:13994;rinstance=7e938a4735992dc7;transport=udp>
Contact: <sip:asterisk@10.8.0.101:5060>
Call-ID: 531afee4691904a8252374cf7cf80db3@10.8.0.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 07 Oct 2015 09:46:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.8.0.7:13994 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.8.0.101:5060;branch=z9hG4bK28967712;rport=5060
Contact: <sip:10.8.0.7:13994>
To: <sip:2000@10.8.0.7:13994;rinstance=7e938a4735992dc7;transport=udp>;tag=8383e30b
From: "asterisk"<sip:asterisk@10.8.0.101>;tag=as2586effb
Call-ID: 531afee4691904a8252374cf7cf80db3@10.8.0.101:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces, eventlist
User-Agent: Bria 3 release 3.5.5 stamp 71238
Allow-Events: hold, talk
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '531afee4691904a8252374cf7cf80db3@10.8.0.101:5060' Method: OPTIONS


Уже данные летают по OpenVPN
порты в Брии ограничены 10000-20000
RTP Starting Port UDP в шлюзе: 9005
Стоит RFC 2833 с обоих сторон.


Уже трясёт от него. Неделю не могу решить проблему.
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ReloadXP
СообщениеДобавлено: Ср окт 07, 2015 13:03 
Не в сети

Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
дамп трафика с порта WAN шлюза снять не могу. Он в бридже. За натом. Могу попробовать настроить микротик на то чтобы он всё мне в WireShark передавал и тем самым будет дамп всего бриджа. Отдельно, как я понимаю это только физически через машину его надо прогонять...
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Evgeniy Ryzhov
СообщениеДобавлено: Ср окт 07, 2015 18:26 
Не в сети
Сотрудник D-LINK
Сотрудник D-LINK

Зарегистрирован: Ср янв 22, 2014 18:37
Сообщений: 1158
Измените дублирующиеся номера в настройках шлюза.
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ReloadXP
СообщениеДобавлено: Ср окт 07, 2015 21:34 
Не в сети

Зарегистрирован: Пн сен 28, 2015 14:49
Сообщений: 11
Evgeniy Ryzhov писал(а):
Измените дублирующиеся номера в настройках шлюза.

Весело. А я считал что если галочка напротив не стоит, то оно неактивно.
Напьюсь сегодня, проблема решена. Спасибо! :D
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