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

[Oliver Brandt]

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