Silva
Envoyé : mardi 10 janvier 2006 22:51
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] codecs order and so on
Doing in the console show translation i can see that it seems not be
possible to translate from any to g729 codec, or from g729 to any. So, let
Title: Message
The problem
:
an asterisk box
with 2 fxo
First fxo just
receive calls from pstn (ulaw)
Second fxo receive
and send call to mobile network thru a sipbox(ulaw)
Calls to pstn are
sent to a pstn provider accepting only g729
Internal calls
doesn't care of
Doing in the console show translation i can see that it seems not be
possible to translate from any to g729 codec, or from g729 to any. So,
let me try to find a reason for this.
When you have first allow=g729 (preferred codec)
all the calls to pstn providers work because the phones and asterisk
That should be controllable by a weight, for example 2 peers:
A -- G729, G711
B -- G711, G729
What's currently happening is that * starts transcoding between the two
(g729 for A and G711 for B), what i would like is to apply a weight to
peer A so that the codec of choise at both sides becomes
The way I said is the "gospel" of how it happens. /bOn Aug 16, 2005, at 1:42 AM, Erik Versaevel wrote:That should be controllable by a weight, for example 2 peers: ___
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I remember many discussions about inteligent codecs negotiation in
asterisk, but seems, however, this isn't as simple to implement as it
looks... :-(
PJ
Erik Versaevel wrote:
That should be controllable by a weight, for example 2 peers:
A -- G729, G711
B -- G711, G729
What's currently
As someone that spent a week or more with anthm refactoring this code I can tell this is how it was when we were done and the code was accepted. So I do know a bit about this area of sip and iax./bOn Aug 16, 2005, at 3:03 PM, Pavel Jezek wrote:I remember many discussions about inteligent codecs
hi,
i have this topology
pstn+(e1)asterisk1-asterisk2-sip client
asterisk1,asterisk2 allow (g729,alaw)
sip client prefer g729, then alaw
can you someone describe codec negotiation when call for sip client arrive
from pstn? (can i set g729 for calls from pstn? )
thanks
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer peer and look for codec order entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not match,
asterisk will
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer peer and look for codec order entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not
Here is an example:
Call comes in via PSTN... ulaw is the native format of the channel.
On the sip side you have g729,ulaw as the codec order. That call
will end up being ulaw because we send the native format as our first
choice above all because we don't want to transcode.
/b
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