Hi Dave! Thank you very much for replying!
> what gateway provider are you referring to? doesn't your sip phone webcalldirect (it does not seam to support iLBC directly) > connect directly to * as your diagram indicated? Yes, my sipphone ist connected directly to * and also the gateway provider is directly connected to *. My * is on a root server at hosting provider (high bandwith internet connection to the gateway provider) but my phone is connected through DSL with a very limited upstream. For this reason I'd like asterisk to do the codec conversion from iLBC to ulaw. I bett all I have to do is load the codec or/and the codec translator for iLBC to ulaw. But when googleing I only find articles the describe, that * is doing the codec translation automatically. I can't find any information on how to load a codec or the translator manually. I'm probably just using the wrong search string in google... When * starts translators are beeing loaded, but as far as I can see non for iLBC to ulaw. I've put together another test setup with to sip phones to clarify the problem: [phone1] disallow=all allow=iLBC [phone2] disallow=all allow=ulaw When calling from one phone to the other I get the following message: chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling call to phone2 Thank you very much again! Oliver _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users