Well does g729 have to run on both legs of a call? For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me.
On 8/28/07, Andres <[EMAIL PROTECTED]> wrote: > > > > >and reload, strange things begin to happen. A show g729 shows this: > >5/0 encoders/decoders of 5 licensed channels are currently in use > > > > > I think you have a loop of some kind. As you can see none of those call > are actually established since no decoders are in use. Try to debug and > see why those 5 calls are acually not connected in the first place. > > >and suddenly I can not hear anything if I try to make a call. From > >observation, it almost seems like other units on the network are using > >the g729 codecs, but doesn't my sip.conf prohibit g729 unless > >expressly allowed?! Why would allowing g729 under one extension allow > >everyone else to suddenly start using g729? > > > >_______________________________________________ > >--Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > >asterisk-users mailing list > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > Andres > http://www.telesip.net > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
