Yes my friend...CONFIRMED!!! G729 on both sides 2010/9/15 Ishfaq Malik <i...@pack-net.co.uk>
> Have you checked that the codec order on the phone matched the order set > on the server? > > On Wed, 2010-09-15 at 17:04 +0200, Danny Dias wrote: > > Hello, > > > > > > I'm having some problems with a total SIP Asterisk scenario, some > > extensions when make internal and outgoing calls can't hear very well > > the other party, not echo, not packet lost....the problem is that the > > volume seems to be very low...what could be happening? i'm not sure > > what to check > > > > > > Thanks! > > > > -- > > Salu2 > > > > > > -- > > _____________________________________________________________________ > > -- 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 > > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _____________________________________________________________________ > -- 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 > -- Salu2
-- _____________________________________________________________________ -- 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