[asterisk-users] Increase Volume - SIP

bilal ghayyad bilmar_gh at yahoo.com
Mon Jan 7 16:31:15 CST 2008


Hi Marc;

Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.

Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are facing a low volume
when you call from FXS (or from IP Phone)?

There is a way to increase the volume on the fxs or
fxo ports by using the following:

modprobe zaptel
modprobe wctdm fxorxgain=12 (or more)

But asterisk should not be running, and you should
unload zaptel and then reload it (as it takes this
argument only when loading), and you need to do it
each time you restart the machine (or it can be
hardcoded if you found a good result).

To unload zaptel, use modprobe - r zaptel and modprobe
-r wctdm (But remember to let asterisk off, not
running).

You can also bypass fxotxgain, also you can use
fxstxgain and fxsrxgain for fxs ports, remember that
here we use fxs for fxs and fxo for fxo where the case
differs in configuring that in zapata.conf

Hope that help and please let me know what happened
with you.

Regards
Bilal


--------------------------------
marcelocbf at comcast.net wrote:

> Can someone tell me if there is a way to increase
the volume of a
 conversation that occurs between two SIP channels or
between a SIP and an
 IAX channel ?
> 
> My headsets are set to the maximum volume but the
voice is still low
 ... I know there is a configuration in zapata.conf
for the digium
 cards, but is there a place I can set this up for RTP
conversations ?

No. Try to adjust the mic volume if you can.

(Alternative solution: Shorter ethernet cables make
the audio
a bit louder. ;-)


Regards,
  Philipp Kempgen


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