thanks for your answer "set your nativeformats to the negotiated codecs and let Asterisk do the rest." would be great for me but I could not find how to do this if transcode is not allowed and one of my codec is compatible with the other side codec I would like to know which codec it is and then set my nativeformats... i don't care about the phase my channel (and the middleware/hardware managed) is flexible.
regards Aurèle 2016-11-21 16:04 GMT+01:00 Joshua Colp <[email protected]>: > On Mon, Nov 21, 2016, at 11:01 AM, Aurele Traynard wrote: > > Hi everyone, > > > > The main goal of the channel is to make or receive call from another > > channel (mainly SIP) > > I writing a custom channel I have now something working with codec "alaw" > > When I add multiple codecs Asterisk's core "negociate" the good codec and > > give it in the "request" function then I can know wich codec I have to > > use. > > (maybe it is not asterisk's core wich do this?) > > When the call comes from my channel, I can'tknwo which codec will be used > > by the other channel... > > > > I tried to read chan_sip and chan_iax2 as I did to write my custom > > channel, > > but I could'n see what to do... > > > > thanks for any help and feel free to ask anything about my problem if I > > was > > not clear enough. > > There is currently no mechanism to know what the other side when > answered has negotiated. It's been talked about previously that it would > be good to have such a thing, but it does not exist currently. > > The only thing you can do is set your nativeformats to the negotiated > codecs and let Asterisk do the rest. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
