OK, it looks like my understanding was way off base.
On the advice of Dave D., Eric V. and Dave W., I've tried: - Locking the phones into G.711 - Increasing the DTMF On/Off times from 50 to 150 (tested at 25ms increments) - Increased the DTMF level from -15db to -7db - Verified that RFC2833Control is enabled - Verified that ViaRTP is enabled - Verified that RFC2833Payload is set to 101 I have two Customers, a Vendor, and my home answering machine that I am using for testing, and I have not been able to find a combination of these parameters that will get DTMF to work reliably. I also think I got lucky on the day that I made a bunch of test calls and came up with my estimate that it was working 75% of the time... I've done more extensive testing since then, and it seems to work correctly more like 10-20% of the time. I've probably made 20-30 phone calls to my test targets just today, and only had a couple successes. Mike Burden Lynk Systems, Inc e-mail: [email protected] Phone: 616-532-4985 From: John Lodden [mailto:[email protected]] Sent: Friday, March 05, 2010 3:40 PM To: Burden, Mike Subject: Re: [sipx-users] DTMF Issues Revisited Mike, MNS is a CLEC with Class 4/5 switches (i.e. we can provide dial-tone just like the LEC). We don't interface at the FXO/FXS level, we have many hundreds of DS3 level SS7 trunks into the PSTN. Most likely the issue is the phone will not send DTMF until "Connected" message is sent back from the far side. -jrl On 3/5/2010 3:27 PM, Burden, Mike wrote: Good afternoon, Most of the time (about 75%) we can call a Customer/Vendor/etc that has an automated attendant, and we are able to make selections or choose extensions using DTMF on the keypad. Once in a while, though, we make a call and are not able to make a selection unless we wait until the auto-attendant on the receiving end finishes its spiel. Since our configuration isn't changing between calls, and I assume that our ITSP isn't changing configurations, I believe that DTMF is configured correctly in sipXecs (or else it wouldn't work right 75% of the time.) My theory is that the ITSP must have a number of FXO/FXS devices to connect VOIP calls into the POTS system. I'm guessing that they may vary somewhat in the quality of the DTMF tones they generate, and that different calls being routed to different FXO/FXS devices explains why sometimes DTMF works fine and sometimes it doesn't. Could I be onto something here, or is there a flaw in my reasoning? Mike Burden Lynk Systems, Inc e-mail: [email protected] Phone: 616-532-4985 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
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