On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan & Company, LLC <[EMAIL PROTECTED]> wrote: > In the sip peer definition, > > disallow=all > allow=g729 > allow=ulaw > > SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw > for the ZAP calls. But, when your polycoms talk with each other, as > long as all necessary REINVITEs happen, they should use the 729 codec I > believe. Remember however, that many options to the Dial application, > like t,w,m,k (or so) REQURE asterisk to remain in the media path. > > moj
AFAICT, I say that in this case this will not work... Very unfortunatelly. It's related to the way the current asterisk versions behave regarding codec negotiation / renegotiation. Your sip.conf entry will have the phone-asterisk leg be g729 and the other leg, to the PSTN, will be a/u-law. When bridging, asterisk is not clever enough (yet!) to renegotiate the SIP leg back to a/u-law and either a) it transcodes or b) the call fails if no transcoder is available... I've given this issue some testing with no sucessful results in the recent past... (check last two/three months list archives) Asterisk really needs a revamped media renegotiation algorithm ! Will we get one in 1.6 ?!... I guess not. Again, unfortunatelly, as this is a very core, very important issue. (feel free to correct me and give me the good news !!!) Cheers, -- exvito _______________________________________________ -- 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
