Asterisk 1.6 / poor hearing, sound loss

Question

Asterisk

pstn city line -> fxo port of the Linksys SPA8800 gateway -> registration via trunk pstn1 on asterisk 1.6 version (asterisknow, centos, on the local network) -> softphone/cisco_SPA330/cisc07930

Everything is fine, everything is fine. It has been working for a month, BUT there is a big minus.
ONLY(!) when outgoing conversation on the city line (for example exten 101 -> 8495777777 ) is bullshit. The interlocutor from the city is hard to hear + a small echo + if I start to say / hiss something - I COMPLETELY interrupt my interlocutor and hear nothing. If you turn off the microphone (softphone, tsiska), then I hear the interlocutor mega perfectly.
In all other cases, everything is fine.

I also connected the bastard gateway dlink 7111 ( 1fxs + 1fxo) + analog telephone (in addition to sip phones), turned off linksus completely (for testing) - VoIP-> PSTN traffic went through the dlink gateway. The situation is similar. I start talking - I immediately stop hearing the interlocutor. With this test, I kind of ruled out the possibility of interference and problems from the linksys gateway.

In general, all 4fxo+4fxs lines on the gateway are used. I use one for testing.

Where to dig I do not know. alaw|ulaw codecs.
What configs, logs, debugs are needed - I can lay out.

I promise to the bravest who can help)

Answer 1

In my memory, such (although I had a communication) were due to:
a) Silence detection - did not always work out correctly and turned the conversation into stuttering, communication failure;
b) enabled ICE in the softphone - in my opinion, it did not affect the quality of the conversation, but only the authorization, but for old times' sake I prefer to turn it off;
c) setting the codec, especially the buffer - negatively affects the call delay but positively affects the sound quality, you need to look for a middle ground;
d) network stability - if there is a large Jitter (ping time spread), then you will not see good quality;
e) codec negotiation - here it is already very similar to your case, with outgoing and incoming calls, the gateway and the server could not normally agree on the codec and different codecs were selected for different situations, which unpredictably affected the quality of the conversation.
You need to look at the SIP logs of the incoming and outgoing calls and compare what the difference is. You can also raise syslog and compare the logs of incoming and outgoing calls on the gateway.

Answer 2

Sounds like an echo cancellation issue. Play around with the echo settings. Use OSLEC as an echo canceller - it's more efficient.

Answer 3

I can try to help. Write to PM.