[asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
> You might also try "canreinvite=no" for both your phone and the sip 
> peer.

Yeah, there is definitely no re-inviting going on.  Both Asterisk and
the local handset are in a local network behind NAT with reference to
the SIP server that requires INFO.

> I think it's normal procedure for Asterisk to drop out of the 
> call path once the call is established between two peers.  The 
> canreinvite directive forces asterisk to remain as an intermediary, and 
> it will probably do the transcoding that way.

Indeed, this is my understanding as well, but I am definitely not
getting a bridging of the sipphone and sip provider through a re-invite.
The NAT would not facilitate it.

> If I'm not mistaken this 
> is also useful for making calls between two system that have no common 
> codecs.

Right.

I need to use ekiga or one of the ATAs here that I know support rfc2833
so that I can eliminate this possible need to transcode inband to
info/rfc2833 in order to narrow down the field.

b.

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