For OH323 there is a workaround
Before dialing out, do in your dialplan :
exten => XXX,1,SetGloabalVar(OH323_OUTCODEC=g729)
We are also preparing a version that has endpoint configuration
like in sip.conf. It will be ready soon.
George
Mike Tkachuk wrote:
Hello,
I fixed this problem for me with some asterisk patching.
You can download patches at b2bua.berlios.de.
Short explanation: new option 'O' in Dial application will send only 1
codec (same as incoming) in outgoing invite. Curently only SIP channel
patched.
P.S. I'm not really good in asterisk internals, so, I'm sorry if this
'ugly' hack, but it works for me fine.
On Sat, 19 Mar 2005 17:01:50 -0700, Kevin P. Fleming
<[EMAIL PROTECTED]> wrote:
Yves wrote:
I receive G729 & G723 calls that I send to a provider who can handle
both too, is it impossible to tell Asterisk to keep using the same codec
for in & out ? It seems that he only follows the codec list in order.
You are correct. Since Asterisk is a UAS/UAC and not a proxy, it
negotiates both sides of the call independently. Given that, when you
define a SIP user as allowing both G.729 and G.723, and the phone offers
both, Asterisk will pick one and move into the dialplan.
However, if your provider's peer entry is also configured to allow both
G.729 and G.723, things should work fine. By default, Asterisk will
force a preference to the codec being used by the existing channel, to
try to avoid transcoding. That means that if you call in using G.729,
and the peer is set to prefer G.723 normally, Asterisk will still list
G.729 first in the outgoing INVITE to the peer. As long as the peer
respects that request, the call should go through without transcoding
needed.
If you are not experiencing this behavior, then you'll need to post more
details and a 'sip debug' trace so we can figure out what's happening.
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