[Asterisk-Users] Problem with echo

Iain Stevenson iain at iainstevenson.com
Wed Jul 2 04:00:21 MST 2003


rxgain and txgain are used, for example with the X100P.  As I understand 
it, the echo problem with a SIP to PSTN implementation in * has two 
components:

- echo resulting from the digital to analogue conversion at the X100P
- acoustic feedback within the handset used

The former is reduced by using the zaptel echo canceller set by this in 
zapata.conf:

echocancel=yes
echocancelwhenbridged=yes

The choice of echo canceller to use is made when you compile zaptel.  mec2 
is the default.  You can enable aggressive cancellation in mec2 but this 
can be a bit too severe making calls sound almost half duplex.  Mec3 seems 
to be a bit unstable.

You can reduce feedback related echo by tuning rxgain and/or txgain.  A 
value of -3.0 will set the gain at about 70% of its initial value.

  Iain




--On Wednesday, July 2, 2003 3:40 am -0700 "Ing. Angel Gomez Garcia" 
<angom at telnor.net> wrote:

>
>     I have a SIP FXO 8 port VoIP gateway, and it has a parameter called
> 'input gain' wich is the one I modified, there might be a similar
> parameter on the configuration for the hardware you are using.
>
> Dan wrote:
>
>> Hi,
>>
>> What do you mean by pstn-gateway?
>> There is no "input gain" parameter in zapata.conf file?
>> It is about "rxgain"?
>>
>> BR,
>> Dan
>>
>> ----- Original Message -----
>> From: "Ing. Angel Gomez Garcia" <angom at telnor.net>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Wednesday, July 02, 2003 11:48 AM
>> Subject: Re: [Asterisk-Users] Problem with echo
>>
>>
>>
>>
>>>    I had a similar problem and solved it changing the params of "input
>>> gain" on my pstn-gateway, change from a value of 10 to a value of 1 and
>>> that eliminated the echo on the SIP Phones.
>>>
>>> Dave Packham wrote:
>>>
>>>
>>>
>>>> Same prob here.   15 SIP phones only get eco when going to the PSTN...
>>>>
>>>> if you find something let me know
>>>>
>>>>
>>>> Dave
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>>> dandre at iris-tech.fr 7/1/2003 8:53:13 AM >>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>> Hello,
>>>>
>>>> I can't have asterisk working without echo when I place a call from IP
>>>>
>>>> phone (SIP or H323) to a PSTN Phone. The called number as no problem
>>>> with echo but there is a very audible echo in the SIP phone. This
>>>> situation occurs either when connected with ISDN card thru i4linux
>>>> driver and with my openline card from voicetronix.
>>>>
>>>> Do you have any suggestion fo that?
>>>>
>>>> Regards,
>>>>
>>>> Daniel ANDRE
>>>>
>>>>
>>>>
>>>>
>>>>
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>>>
>>>
>>
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>
>
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