[asterisk-users] Early codec selection / negotiation

[Francois Marier]

Hi Faisal,

Thanks for your note.

On 2011-03-06 at 17:44:35, Faisal Hanif wrote:
> If you dialout call without answering and allow all codec for both peers
> then codec negotiation will be direct between endpoints and asterisk will
> only do media pass-through.

Now, if I understand you correctly, you are saying that my dialplan:

  exten => _1800NXXXXXX,1,Set(CALLERPRES()=allowed)
  exten => _1800NXXXXXX,n,Set(CALLERID(all)=Francois Marier <12345>)
  exten => _1800NXXXXXX,n,Dial(IAX2/username:password at pstnpeer/${EXTEN})

  exten => 1000,1,Dial(IAX2/guest at asteriskpeer/123)

is causing my asterisk box to "answer" the calls before passing them on?

I guess I'm not quite sure what you mean by "dialout call without
answering".

Does it have to do with the "canreinvite" or "directrtpsetup" in sip.conf?

Cheers,
Francois

-- 
Francois Marier                         identi.ca/fmarier
http://feeding.cloud.geek.nz          twitter.com/fmarier