Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw

[dave cantera]

oliver, ugh,  it is too obvious... why did it take me so long to figure it out... both phones have to have to negotiate the same codec for audio...  as far as I know, *  is supposed to do automatic translation and your gateway should be doing translations only on the below codecs.   I haven't had that experience yet... one phone may be connected to your * box, but your other phone is *not*  connected to *.     it is connected to a voip provider...  since they don't do any translation other than below.  the * connection to webcalldirect must have one of these codecs in the sip.conf for that extension, the extension where webcalldirect is coming in, that is... phoneX -----> * ---------> webcalldirect -----> phoneY which one is phone1 and which is phone2? phoneX<---- * <---------- webcalldirect <-------phoneY -----------------| -------------------------| ----------------- local LAN                  Internet                        local LAN some code                no codec control                no codec control control                      little or no call quality control the phone connected to * will also select a code that matches up with the caller (webcalldirect)...   you have no advantage whether or not * converts the audio to the phone connected to *.    you won't get any better reception from webcalldirect because you are not changing that connection. also,  I would change iLBC to ilbc,  case may make a difference...  don't know for sure...  perhaps someone else does... hope that is clearer... daveC Codecs G.711 (64 kbps) G.726 (32 kbps) G.729 (8 kbps) G.723 (5.3 & 6.3 kbps) GSMFR (13.2 kbps) Temporarily unavailable due to technical difficulties. Oliver Brandt wrote: 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 -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000

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