[asterisk-users] DTMF Problem w/ MeetMe

[David Backeberg]

On Thu, Jun 4, 2009 at 9:34 PM, Phillip Heller <pheller at me.com> wrote:
> Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco
> 2821(CME,Europe) <-SIP-> Asterisk(Boston)
> debugging enabled on Asterisk, I see that I often get duplicate DTMF
> entries.  So where I might have dialed 1234#, Asterisk sees 112344# or
> similar, under scenario 2.
> Any suggestions?

Yes.

Why on earth do you send the call to Europe and back? Can you leave it
in Boston?

Question. Have you considered flashing the 7941s with the SIP
firmware? As it stands right now, your voip gateways have to transcode
the audio to Skinny to/from SIP.

Question. Why not take some steps out of the equation, and terminate
the SIP straight into asterisk?

Question. What good is the Cisco gear doing in your diagram?

Question. Assuming you ever get through, what does the audio sound
like? Are your dropping packets and having the voice sound like
garbage?

Another suggestion: you can debug on the Cisco side too. The Cisco
gear has a lot of choices to tweak DTMF, including sending it inband
as audio tones. If you haven't already, get the thick Cisco manual
that shows all the choices for DTMF that correspond to your gear.