>
>> 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
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