[asterisk-users] Odd DTMF problem when receiving calls

[David Backeberg]

On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie
<me at chrishowie.com> wrote:
> I've been up and down this issue for a few hours and I cannot for the
> life of me determine why simply defining a peer causes Asterisk to offer
> telephone-event.  I have tried specifying dtmfmode=rfc2833 or
> dtmfmode=auto in [global] and neither change has any effect.  As I said
> above, I've copied every configuration directive in [global] into the
> peer definition for the company PBX, and calls still work.
>
> So I'm at a loss to explain this.  The problem does not seem to stem
> from my configuration, but I'm not entirely sure what else could be the
> problem... an Asterisk bug perhaps?  I don't want to jump to that
> conclusion since this is my first day tinkering with the software.
> Perhaps someone more knowledgeable can steer me in the right direction?

Wheee. You don't say anything about what 'company PBX' is, so we just
have to guess based on your description. Based on your description,
your 'company PBX' requires that the endpoints it communicates be
registered before-hand. Having a definition for the sip peer in
asterisk makes asterisk continually register with the 'company PBX'.

So it is not necessarily your case that you think it is, that rfc2833
is required, but rather that for 'company PBX', any sip endpoint must
first be registered. Both asterisk and 'company PBX' probably support
large numbers of possible DTMF or signaling possibilities, and it's
not surprising that you can get away with several possible values.

If you do 'sip show peers', with and without the config in your
sip.conf (use ; to comment it out and 'sip reload' to commit your
changes), you should be able to verify that THIS is the real problem.

And no, this is not an asterisk bug.