I got some advice from Josh Colp that has helped with some of my problem:
it may have a little logic flaw in the way transcoding is supposed to be done, from the way your message is I would say you are getting hit by this. (Upgrading to latest CVS head will fix it)… but one solution is to be the following in asterisk.conf in /etc/asterisk
[options]
transcode_via_sln = no
That’ll cause it to bridge the two and not try to transcode through signed linear. Enjoy!
Well that worked, after a fashion. Now AS LONG AS I ONLY USE G.729 ONLY things are fine.
But the 841 does all kinds of codecs, and so I'd like it to use g.726 to talk to a provider that doesn't speak g.729. So I set the sip.conf for the phone to "disallow=all; allow=g726,g729" and then try to connect to the g726-only server:
Apr 25 00:58:42 NOTICE[5839]: channel.c:1833 set_format: Unable to find a path from g729 to g726
After playing with this for as long as I could stand to, it appears that IFF I am talking to a g729-only endpoint and I set the SIP phone to use g729 only, things are fine.
Once I deviate from that (unfortunately restrictive) setup, I can't seem to do anything. In other words, if g729 is in the mix it seems to always choose it despite my preferences, and things get hosed.
I'd love to hear from someone who has conquered this.
Thanks.
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