bilal ghayyad
2008-Jan-07 22:28 UTC
[asterisk-users] asterisk-users] Increase Volume - SIP
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 increasethe 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 thevoice 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 ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs