[asterisk-users] audio gain in SIP channel

[Rafael dos Santos Saraiva]

I dont using these functions (AGC/ DENOISE). My suggestion... try invert
the priorities:
Set(DENOISE(tx)=on)
Set(DENOISE(rx)=on)
Set(AGC(rx)=xxxx)
Set(AGC(rx)=xxxx)

And try higher values.. is more easy the perception if the values are
larger than default.


Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>


2014-07-24 9:23 GMT-03:00 lore <pino.maiuli at gmail.com>:

> thanks a lot Rafael.
> could you tell me also something about AGC(rx)=xxxx?
> I mean, i've tryed
>
> Set(AGC(rx)=xxxx)
> Set(AGC(rx)=xxxx)
> Set(DENOISE(tx)=on)
> Set(DENOISE(rx)=on)
>
> using xxxx=8000, 16000 and 32000 but all calls looked like to have se same
> audio gain.
>
> thanks for your rapid reply.
>
>
> 2014-07-24 13:12 GMT+02:00 Rafael dos Santos Saraiva <rafaelsnsa at gmail.com
> >:
>
>> Hi
>>
>> To using VOLUME function the syntax is:
>> Set(VOLUME(rx)=+n)
>> Set(VOLUME(rx)=-n)
>> Set(VOLUME(tx)=+n)
>> Set(VOLUME(tx)=-n)
>>
>> I think is not possible retrieve the value of the channel.
>>
>>
>>
>>
>> Att,
>> *Rafael dos Santos Saraiva*
>>  <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
>>
>>
>> 2014-07-24 7:52 GMT-03:00 lore <pino.maiuli at gmail.com>:
>>
>>>  hello all,
>>> i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
>>> using functions
>>> AGC and VOLUME, but seems that does not work at all.
>>> There is a way to check this values during setup/call?
>>> Maybe is it not possible realize what i'd like to do?
>>>
>>> Could anyone can help me on this?
>>>
>>> thanks a lot in advance
>>>
>>> regards
>>>
>>> Lorenzo
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>                http://www.asterisk.org/hello
>>>
>>> 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 --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> "Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non
> conta un cazzo, 1941 ... sono anche un autore"
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140724/8e8c9d31/attachment.html>