[asterisk-users] SIP to IAX2 with delayed echo

[Dave Fullerton]

There are also settings which will turn on local echo cancellation for 
the handset, headset and/or speaker phone. I don't recall their names at 
the moment. They are off by default on the handset and headset unless 
you're using a very recent (3.0+) SIP app.

Tim Nelson wrote:
> I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high.
> 
> You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-)
> 
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
> 
> ----- "c james" <cjames at callone.net> wrote:
> 
>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>> having a conversation.  Call quality is reported as good except for
>> an
>> echo with a 3 second delay.
>>
>> Most of my searches are saying echo happens only on the PSTN piece,
>> but
>> there isn't one here.
>>
>> Can someone point me in the right direction?
>>
>> Asterisk 1.4.21.2
>> Under 40 users
>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what
>> they wanted to use!)
>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>