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Vladislav Kolotsei2015-09-14 21:28:20
Asterisk
Vladislav Kolotsei, 2015-09-14 21:28:20

Caller inaudibility (or call rejection)?

I need help in solving a problem that is already haunting me. I don't understand what could be happening. The sound in the call constantly disappears. And this can happen, either when the operator calls, or when he is called. After listening to the recordings of conversations, I realized that the sound disappears precisely from the subscriber with whom the operator communicates. I concluded that the problem is in the Internet connection between the Openvox gateway <-> Router <-> Asterisk server. I wanted to ask you for help, how can I test this direction? How can you find out exactly what the problem is? Can you give any recommendations?
I'll drop the session of the call that failed, which I analyzed using Wireshark.
da4ed2de8457b92c.png
It shows that everything is in order with the call. I would also like to try to drive the rtp session somehow into wireshark, but I don’t know how to do it
PS I drove the RTP session into wireshark and saw the following (below is the picture of the call where the voice disappeared):
c0060131e5fc45b084a376956a2af633.PNG
Statistics were made between the Openvox gateway and the asterisk server. It turns out there are no losses. But if you listen to the recording of the conversation, you can hear that the sound has disappeared from the subscriber's side. It turns out that he did not reach the asterisk server. What can be the conclusion? What is the gateway itself faulty?

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4 answer(s)
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catHD, 2015-09-15
@catHD

If you didn’t write a dump on 1 port ( 5060 ), then there is RTP in the dump, do:
- Telephony
- RTP
- Show All Stream
? Is there a pattern? 30 sec?

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Armenian Radio, 2015-09-14
@gbg

The RTP session in wireshark should be visible (if it sees SIP). You can even listen to it.

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Vitalik, 2015-09-15
@pvs

You have already been told what to do forum.asterisk.ru/viewtopic.php?f=3&t=6731 and where to look.
If you tried all this and did not help, then put a dump on the server where Aster is standing
+ enable logging on Asterisk, and then analyze the collected information.

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solalex, 2015-09-15
@solalex

I would recommend looking at the trunk codec settings on Openvox

disallow=all
allow=alaw

instead of alaw, the codec on which the sound exactly passes.

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