asterisk users - Jun 2009 - DTMF Problem w/ MeetMe

First, the scenarios:

Call placed from Boston to locally configured Asterisk Meetme extension:

Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston)

Call placed from Boston to European Asterisk Meetme extension:

Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco  
2821(CME,Europe) <-SIP-> Asterisk(Boston)

In the 1st scenario, everything works great.  I dial the extension on  
the 7941, Asterisk parses DTMF just fine.

In the 2nd scenario, DTMF digits do not interrupt the "Please enter  
conf #" announcement, and when I do finish entering the same conf #  
used for scenario #1, it announces "Invalid Conf #".  With channel  
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.

All dial-peers involved on the 2821's are configured for dtmf-relay  
rtp-nte, and debugging shows this to be the negotiated dtmf method.

Interestingly, there is a 3rd scenario which adds to the confusion:

Call placed from Boston to European Asterisk Meetme *DID*:  (note the  
trip out to the PSTN is simply a hairpin in this case)

Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco  
2821(CME,Europe) <-PRI-> PSTN <-PRI-> Cisco 2821(CME,Europe)
<-SIP->
Asterisk(Boston)

This is using Asterisk 1.6.0.5

I've tried altering relaxdtmf in sip.conf to no avail.  dtmfmode is  
set to auto.

Any suggestions?

Regards,

--phil

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.