[asterisk-users] Odd DTMF problem when receiving calls

[Christopher David Howie]

Thanks for the reply David, this clarifies a lot for me.

On 01/10/2012 08:28 PM, David Backeberg wrote:
> 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'.

The company's PBX is ININ (Interactive Intelligence IC).

> 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.

I'm fairly certain that RFC2833 is required, as the client software
clearly indicates that the connection was dropped due to lack of RFC2833
support:

17:53:23: Initializing
17:53:23: Sent to user Chris.Howie
17:53:23: Outbound Call:  me at myserver.example.com
17:53:23: Dialing
17:53:26: Disconnected [RFC 2833 Required]

It appears that, by registering, Asterisk is able to ascertain what
capabilities the peer supports and wants, and so it knows to offer the
telephone-event stream to incoming calls.  Since the SIP dialog differs
in how Asterisk responds to the incoming call (it offers telephone-event
only when the peer is defined in sip.conf), this seems like the case to
me.  Does that seem reasonable, or am I missing a piece of the puzzle?

If this is the case, how would I get Asterisk to offer telephone-event
on incoming calls without having to register every ININ-based PBX that
might call me?  (This is more theoretical than practical, as I doubt
that any ININ PBX is going to call me over SIP.)

> 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.

One of the things I discovered during testing is that if I remove the
peer and issue "sip reload" then incoming calls are still accepted.  I
have to completely restart the Asterisk daemon to get it to forget about
the peer definition.

> And no, this is not an asterisk bug.

I suspected that it wasn't, which is why I asked here first instead of
using the bug tracker.  :)

-- 
Chris Howie
http://www.chrishowie.com
http://en.wikipedia.org/wiki/User:Crazycomputers

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