[Asterisk-Users] Re: Asterisk Conferencing using g729

[Chris A. Icide]

On 12:53 AM 9/7/2004, Tony Mountifield wrote:
 >
 >> Another related question: Is there a way to just use g729 for the 
conference and for nothing
 >> else. The problem I have is that I have Broadvoice ( BV rocks, by the 
way) which requires
 >> ULAW and sends DTMF inband. If I allow g729 in the sip.conf, Asterisk 
complains that inband
 >> dtmf is only supported under ULAW and incoming dtmf does not work 
through Asterisk,
 >> something I must have.
 >

You may very well have hit on a bug (well, really a feature 
request).  Asterisk *SHOULD* do the conversion, so if your Sip UA 
originates the call, and between it and Asterisk, they choose g729, and 
then asterisk originates a call to Broadvoice, with the selected codes as 
Ulaw, then Asterisk should take incoming out of band dtmf from the UA and 
generate in-band dtmf for broadvoice.

If this isn't happening, it should.

You might be able to reset the codec by using the channel variable for the 
codec.  I forget off the top of my head what it is, but surely the wiki or 
the archives of this list will provide that information.  I'm not sure 
though, once the invite from the UA is sent and accepted, that changing 
that variable will force a re-negotiation of the codec?  If it does, then 
you could force ulaw to the UA when you end up in the broadvoice context, 
or force g729 when going to conference, and default to ulaw....

-Chris