[Asterisk-Users] SIP and ECHO

[Daniel ANDRE]

Hello,

Brian West a écrit:

>I get no echo on mine.. but you can check to make sure your line isn't
>reversed.  A reverse wired jack can do that.
>
I don't think so but I have tested reversed and it doesn't solve my echo 
problem

Daniel

>
>bkw
>
>On Thu, 28 Aug 2003, Brian J. Schrock wrote:
>
>  
>
>>I can minimize doing those tricks, but I cannot seem to get it to go
>>away.
>>
>>On Thu, 2003-08-28 at 11:33, Dan wrote:
>>    
>>
>>>----- Original Message -----
>>>From: "Brian J. Schrock" <brians at anistonetech.com>
>>>To: <asterisk-users at lists.digium.com>
>>>Sent: Thursday, August 28, 2003 6:16 PM
>>>Subject: [Asterisk-Users] SIP and ECHO
>>>
>>>
>>>      
>>>
>>>>Hello,
>>>>
>>>>I have read the information on echo and SIP in the FAQ and I have
>>>>scoured the mailing list for possible solutions, but as yet I have not
>>>>been able to get rid of this echo.
>>>>
>>>>I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed
>>>>into an asterisk server. If I call between the Sip Phone
>>>>(Budgettone-100) and the 4 FXS ports everything sounds great. If I call
>>>>out to the PSTN through the FXO cards I get horrible echo, I have even
>>>>been able when talking loud enough to get a horrible feedback loop
>>>>going. I have tried 4 different echo cancellers in the Makefile for the
>>>>Zap drivers and nonoe of them changed the situation.
>>>>
>>>>I have echocancel = (Any where from 1 - 256, I have tried alot of
>>>>different values), and I have echocanelwhenbridged = yes.I only hear the
>>>>echo start when the call gets bridged onto the outgoing PSTN lines.
>>>>
>>>>Is there anything I can do?
>>>>
>>>>Brian J. Schrock
>>>>
>>>>        
>>>>
>>>Hi,
>>>
>>>For me:
>>>
>>>rxgain=0.8
>>>txgain=0.8
>>>
>>>in zapata conf do the trick.
>>>Now the echo is allmost inexistant. Maybe the sound is not very strong but
>>>the quality is very good.
>>>I have the default echo canceller (no modification in the source files).
>>>
>>>Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711),
>>>Cisco 79x0) and one X100P card.
>>>
>>>BR,
>>>Dan
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>      
>>>
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>>    
>>
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>
>  
>

-- 
Daniel ANDRE (mailto:dandre at iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com

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