[cisco-voip] SIP Trunk question

[Nick Matthews]

95% of the time these are dial peer misconfigurations.  Run 'debug voip
dialpeer' in your two situations with no other calls on the router, and see
if there is a difference.  If you hit dial peer 0, you need to read about
incoming dial peers and fix the problem. Another 3% are just a interop issue
and you need to reconfigure the DTMF method from end-to-end, or somewhere in
the middle.

In very rare situations timing issues prevent DTMF from happening.  Changing
H.323 to slow-start to fast-start or vice versa will sometimes fix that, or
using an MTP on CUCM.

-nick

On Mon, Sep 27, 2010 at 6:40 PM, Eric Brander <mailinglists at rednarb.com>wrote:

> On Mon, Sep 27, 2010 at 2:53 PM, Jim McBurnett <jim at tgasolutions.com>
> wrote:
> > So does anyone have a good idea as to why some calls on a SIP trunk will
> > work fine with DTMF and a few fail?
> >
> >
> >
> > IE—
> >
> > Call to X – AA options don’t work..
> >
> > Call to Y—work fine..
> >
>
> I'm fighting with sometjhing similar right now in a configuration
> between Cisco and NobleSystems (dialer platform based on Asterisk). If
> the call placed from my Cisco desk phone communicates directly to a
> call manager which then in turn passes the data across the SIP trunk
> to Noble then everything works fine. Sometimes however the call will
> be handled initially by the call manager but then the phone
> communicates directly to the Noble SIP server instead of using the
> call manager as an intermediary - when this happens it appears not all
> commands are accepted, including DTMF. It seems load related - the
> more SIP calls in-place, the more likely for this direct connection
> problem.
>
> I can't for the life of me find a solution... but not sure if its
> related to your issue or not but sure sounds similar.
>
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