[asterisk-users] DTMF transmission problem

Shaun Ruffell
Thu, 02 Aug 2012 15:05:49 -0000

On Thu, Aug 02, 2012 at 12:45:28PM +0000, Noah Engelberth
wrote:
> I am having difficulties with customer-bound DTMF being very short
> & clipped off (and basically unusable, as systems on the customer
> side aren't recognizing the DTMF digits, and I can barely tell
> that DTMF is there when I listen on a handset).
>
> My system set up as follows:
>
> PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE

[snip]

> ... Vocal call  quality is fine, DTMF is fine from the customer to the
PSTN, but
> DTMF from the PSTN to the customer isn't ...

 [snip]

> The same symptoms persist whether the PSTN or the CPE initiate the call.

What is the dtmf mode of Metaswitch in the above diagram? Is it
possible that it's muting the DTMF and then not generating the
corresponding DTMF event messages?  Everytime I've seen "clipped"
DTMF in the past it was due to imperfect muting at the PSTN -> SIP
interface.

You should be able to take a packet trace on the interface of the
Asterisk server communicating with the Metaswitch to determine
whether the problem first appears at the switch or in your Asterisk
server.

Cheers,
Shaun

--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org