On Tue, Jun 29, 2010 at 6:42 PM, Philipp von Klitzing <[email protected]> wrote: > Hi! > >> Because the codec is already chosen before the call is made, and you >> told it that g722 is permitted. >> >> There are all sorts of discussions in play about codec negotiation, >> but at this point in time, if you want different behaviour you'll need to >> go and code it yourself > > Look at the list archive - there is a codec negotiation patch around: > > http://lists.digium.com/pipermail/asterisk-users/2010- > February/244835.html > > The OP might also want to consider to use different lines to the same > PBX, one for normal narrowband, and another one for g722. > > Philipp > > > --
Thanks! I'm going to try setting the _SIP_CODEC variable for outbound calls to force ulaw. This should solve the issue. Having two lines would work but I can't sell this to a customer. There has got to be a better way to have Asterisk handle this. With Asterisk in the middle of the RTP stream it knows what both parties support. If it turns out Asterisk is transcoding it could check for a common codec and renegotiate one endpoint. Ryan -- _____________________________________________________________________ -- 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
