[Asterisk-Dev] Codec not negotiating

Kevin P. Fleming kpfleming at digium.com
Mon Apr 4 14:09:47 MST 2005


Clay Reiche wrote:
> I set up the multiple peers and that is a good enough fix for now. Thank you!
> Now, how would I do this if my SIP device was smart enough to determine the
> call type on the fly? Ie...voice(g729) or fax(ulaw). The codec being sent by
> my SIP device(peer) would change depending on the call type. This multiple
> peer solution would no longer work.

That is also an issue; the patch I'm working on will allow you to learn 
(in the dialplan) what codec the calling channel is using.



More information about the asterisk-dev mailing list