[asterisk-users] Zap Codec

[Ex Vito]

On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan & Company, LLC
<mojo at horanappraisals.com> wrote:
> In the sip peer definition,
>
>  disallow=all
>  allow=g729
>  allow=ulaw
>
>  SHOULD work.  Asterisk can't transcode g729, so it should fall on ulaw
>  for the ZAP calls.  But, when your polycoms talk with each other, as
>  long as all necessary REINVITEs happen, they should use the 729 codec I
>  believe.  Remember however, that many options to the Dial application,
>  like t,w,m,k (or so)  REQURE asterisk to remain in the media path.
>
>  moj

  AFAICT, I say that in this case this will not work... Very unfortunatelly.
  It's related to the way the current asterisk versions behave regarding
  codec negotiation / renegotiation.

  Your sip.conf entry will have the phone-asterisk leg be g729 and the other
  leg, to the PSTN, will be a/u-law. When bridging, asterisk is not clever
  enough (yet!) to renegotiate the SIP leg back to a/u-law and either a)
  it transcodes or b) the call fails if no transcoder is available...

  I've given this issue some testing with no sucessful results in the
  recent past... (check last two/three months list archives)
  Asterisk really needs a revamped media renegotiation algorithm !

  Will we get one in 1.6 ?!... I guess not. Again, unfortunatelly, as this
  is a very core, very important issue. (feel free to correct me and give
  me the good news !!!)

  Cheers,
--
  exvito