If I understood your initial objective correctly (and I may not have),
the user's phones are negotiating the codec to be used for each rtp session.
Asterisk parameters can be used to dictate rtp sessions between the sip
phone and asterisk, but that won't influence the next step in which the
If I understood your initial objective correctly (and I may not have),
the user's phones are negotiating the codec to be used for each rtp session.
Asterisk parameters can be used to dictate rtp sessions between the sip
phone and asterisk, but that won't influence the next step in
Did you try having two sip.conf entries for your gateway? Forcing one
with G729 and the other with ulaw? You would obviously need to change
your dialplan accordingly and have each phone configured so that it
would take the proper extension. I have not tried this, it is just
really an idea...
On 07:01 AM 6/24/2004, Rich Adamson wrote:
Now I better understand what you're trying to do.
I'm not a programmer, but I'm fairly certain that you can't dynamically
change codec preference within asterisk on a per call basis. However,
just as soon as this gets posted, someone will likely jump all
On Thu, Jun 24, 2004 at 09:52:45AM -0700, Chris A. Icide wrote:
It sounds like what you are looking for is an Asterisk-wide (or perhaps
channel-specific) preserve_codec option. Where preserve_codec=1 means that
asterisk tries to preserve the originating codec if at all possible, and
