Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.
Hi Paul There are two settings in zapata.conf called txgain and rxgain. You can set these to adjust the volume on your PSTN lines. They can be set in db or as a percentage. Garth --- Paul Goodyear <pgudge@gmail.com> wrote:> Sip to sip calls are fine, both local on Asterisk and over a SIP > gateway, however some people who call on the PSTN line say we are very > queit and vice versa, can the volume be turned up on the PSTN line? > > The volume buttons on the VoIP phones only turns up the others voice, > so this is a fix for us, but how do I make our voices louder for the > people on the PSTN line? > > Thanks. > > Paul. > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I thought the txgain, rxgain was purely for echo settings. Is there a rough guide to this process, or is it a simple case of changing values and testing them? Thanks. On 9/15/05, garth@bitco.co.za <garth@bitco.co.za> wrote:> Hi Paul > > There are two settings in zapata.conf called txgain and rxgain. You can set > these to adjust the volume on your PSTN lines. They can be set in db or as a > percentage. > > Garth > > > > > --- Paul Goodyear <pgudge@gmail.com> wrote: > > Sip to sip calls are fine, both local on Asterisk and over a SIP > > gateway, however some people who call on the PSTN line say we are very > > queit and vice versa, can the volume be turned up on the PSTN line? > > > > The volume buttons on the VoIP phones only turns up the others voice, > > so this is a fix for us, but how do I make our voices louder for the > > people on the PSTN line? > > > > Thanks. > > > > Paul. > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I just did it by ear. Got it right in less than 5 minutes. --- Paul Goodyear <pgudge@gmail.com> wrote:> I thought the txgain, rxgain was purely for echo settings. > > Is there a rough guide to this process, or is it a simple case of > changing values and testing them? > > Thanks. > > On 9/15/05, garth@bitco.co.za <garth@bitco.co.za> wrote: > > Hi Paul > > > > There are two settings in zapata.conf called txgain and rxgain. You can > set > > these to adjust the volume on your PSTN lines. They can be set in db or > as a > > percentage. > > > > Garth > > > > > > > > > > --- Paul Goodyear <pgudge@gmail.com> wrote: > > > Sip to sip calls are fine, both local on Asterisk and over a SIP > > > gateway, however some people who call on the PSTN line say we are very > > > queit and vice versa, can the volume be turned up on the PSTN line? > > > > > > The volume buttons on the VoIP phones only turns up the others voice, > > > so this is a fix for us, but how do I make our voices louder for the > > > people on the PSTN line? > > > > > > Thanks. > > > > > > Paul. > > > _______________________________________________ > > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hello all I have a question about Grandstream HandyTone 386 can Grandstream HandyTone 386 make 2 sim. calls with g729 codec in same time Iyi Calismalar. Ugur GUNCER -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 4548 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050915/248db48c/smime.bin