[Asterisk-Users] Getting DTMF to work with SIP?

[beonice]

Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context. 

My problem is that my overseas DID through Libretel
seems  to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I try to
dial an extension! It plays the welcome message,
waits, and then times out (and hangs up nicely, yes).

I've actually tried setting the dtmfmode parameter in
the [general] context, obviously with no success.

Also, I note that when the incoming call uses SIP, a
physical hangup by the caller is not recognised ... it
basically goes through my timeout routine and hangs up
at some point.

The same context in extensions.conf can be reached
through my California DID which uses IAX, and in that
case I have no problem with DTMF or hangup.

Any ideas on how to solve this? I've got a question
open at Libretel, too, but was hoping someone here may
have already dealt with this?

Thanks,
Maya



		
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