Yes it might be dumb but since asterisk is a pbx and not a sip proxy
it has to perform many other functions as well . But i do think that
asterisk should act little smart in this case
SIP wrote:
> That just seems really, REALLY dumb for a program of this magnitude.
>
> I know this has been patc
has anybody made a patch for asterisk 1.4*?
On 6/4/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
Asterisk by default uses the codec preferred by other device/client .
Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough
to check if it can avoid transcoding by forcing same cod
some work has been done here:
http://bugs.digium.com/view.php?id=4825
but seems to be quite death and probably not directly applicable to
current asterisk src :'(
SIP wrote:
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by on
That just seems really, REALLY dumb for a program of this magnitude.
I know this has been patched here and there by one person or another,
but does anyone know if any of these patches to make CODEC negotiation
actually, you know, negotiate a CODEC will ever make it into the core src?
Jaswind
Asterisk by default uses the codec preferred by other device/client .
Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough
to check if it can avoid transcoding by forcing same codec on other
side of conversation . If both sides prefer g729 then asterisk does
not do transcoding bu
Does anybody has any documentation on codec negotiation within asterisk?
Well im using free g729 codec for testing purposes. i mentioned g729 just as
an example. whatever codec is mentioned in user perefernce, asterisk uses
ulaw to throw out the call.
On 5/30/07, Marco Mouta <[EMAIL PROTECTED]>
so you r sure you have g729 licences installed and ur * is transcoding your
RTP streaming?
Test the work flow with disallow=all and allow=g729, can be my mistake but I
remember to read somewhere on the net any issue about codec negotiating
precedence when you use allow=all.
good luck
On 5/30/07
Hi all,
My configuration is:
USER (connects to)> ASTERISK---(connects to)--->CARRIER-OUT
i want the user preffered codec to pass thru asterisk to carrier-out. what i
mean is:
USER (user uses g729)> ASTERISK---(asterisk should use g729 for
dialing out)--->CARRIER-OUT
instead, this
