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Asterisk + Kerio Control?
Good day!
The asterisk has a NAT + external IP setting. Branches are also behind Nat, there are ordinary routers. On the screen are the standard rules. Phones are registered, the connection is crystal clear. BUT the problem is that the phone at the branch starts RINGING after 15 seconds (if you call through the provider). Everything is fine with the internals.
In a sense, the incoming number is immediately displayed on the screen at the time of the call, the indicator flashes, you can immediately pick up the phone and talk without waiting for it to ring. Relevant for any provider (checked for 3x).
If the same asterisk does not go through Kerio, but stupidly with an external IP address, then everything works as it should.
I suspect that for the Yealink T19E2 phone, some other port needs to be forwarded. But what?
This topic is a continuation of Update from 09/28/2016How to suppress echo on Asterisk?
UPDATE 10/14/2016
1) I assembled a server for a test asterisk to enable debugging and see what happens during a call, because on the main asterisk, with a mountain of providers connected to it, it is extremely difficult to figure it out (if you can filter by a specific provider and phone, please give a comment, I think it will come in handy not only for me)
2) I created one provider and one phone account on the test asterisk
3) I created separate rules for kerio (identical to those indicated in the top of the topic, the only difference is in the local IP of the test asterisk)
4) Ibid. , in the server room I took a separate provider cord with a white IP (this IP is registered in Kerio in the rule for incoming connections), as in all branches, and connected it to a regular router
5) I connected an IP phone to this router
6) The phone received an IP via dhcp from the router and registered on a test asterisk
7) I called this phone through a registered provider I
was already going to wait 15 seconds before it rings, BUT the phone made a sound about an incoming call instantly.
Now I will compare the logs and post later.
Conclusion: I simulated a branch in my server room. The settings are identical, the result is different. Why is this happening?
Perhaps the reason is that conversations are being recorded. And while the record is going on the lagana Internet channel at the branch, the phone does not ring .. Strange. If anyone has an idea what the problem is, please comment.
Another difference is that the main asterisk is on Hyper-V, and the test one is on bare metal
UPDATE 10/18/2016
Logs with a problematic PBX
There are suspicions that recording starts at this moment and the phone rings with a delay:
Scheduling destruction of SIP dialog '[email protected]:5060' in 8896 ms (Method: INVITE)
Scheduling destruction of SIP dialog '[email protected]:5060' in 8960 ms (Method: INVITE)
I'll turn off call recording on Thursday and check if the problem persists
All logs:
UPDATE 10/24/2016
Disabling call recording did not help
UPDATE 10/26 2016
REPAIRED!!!11
In the settings of the Yealink T19E2 phone itself (in the "account" -> "Advanced" tab), disabled the "Distinctive Ring Tones" function and everything rustled!
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В настройках самого телефона Yealink T19E2 (во вкладке "аккаунт" -> "Расшиненные") отключил функцию "Distinctive Ring Tones" и все зашуршало!
Спасибо всем кто помогал разобраться!
У абонентских аппаратов или глобально в Астериск у вас прописано так?
nat=force_rport,comedia
directmedia=no
directrtpsetup=no
Вы наверно не понимаете о чем вы спрашиваете.
Сигнализация ходит по порту 5060 (SIP), голос по портам RTP - для астера это 10000-20000
Сигнализация - это как раза и есть пакет со строками "Ring, Ringing, Try, Busy, Ack, Hangup" и прочими.
Вам надо снять дамп звонка, по SIP протоколу, для внутреннего и для внешнего и сравнить в чем разница.
Тогда поймете почему телефоны на 15 сек тупят.
Интуиция мне подсказывает еще порыть в сторону периода регистраций и qualify. Но это так сказать - просто посмотреть что там в них.
Основное - дампы снимайте и сравнивайте.
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