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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users