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@pstnpeer/${EXTEN})
exten => 1000,1,Dial(IAX2/guest@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
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