Ok trying this again... is there anyone using the SPA-3000 with * ???? I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a completed call, and pressing a DTMF button on the opposing phone results in an audible click and very little if any audible DTMF energy being heard. What is muting the DTMF??? Does * have anything to do with this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
try setting dtmf playback length to .5 in the admin section of the Sipura and try again. On 6/6/06, Doug Crompton <doug@crompton.com> wrote:> Ok trying this again... is there anyone using the SPA-3000 with * ???? > I am not sure if this is a specific problem to it or not. This is > something I really need to fix!!! > > When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot > access (reliably) DTMF menus at the called party, after call completion. > Dialing DTMF is fine. > > I checked by calling myself. Listening to either end on a completed call, > and pressing a DTMF button on the opposing phone results in an audible > click and very little if any audible DTMF energy being heard. > > What is muting the DTMF??? Does * have anything to do with this? I am > not using any dial flags. > > I tried 'inband' in all places with no difference. At one point this > seemed like a * feature problem and I thought removing dial flags fixed > it but that does not now seem to be the case. > > How can I fix this??? > > Doug > > **************************** > * Doug Crompton * > * Richboro, PA 18954 * > * 215-431-6307 * > * * > * doug@crompton.com * > * http://www.crompton.com * > **************************** > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856
Ok well I am not crazy! This seems like such an important issue I am not sure why it has lasted for so long. DTMF is the backbone of everything we do here. Without it we would not have calls!! At least get the DTMF stuff right. I feel a little guilty complaining since this is free but it is also used in some high level demanding situations (not mine) and thus is not a trivial issue. Doug On Tue, 6 Jun 2006, Don Pobanz wrote:> > The other gotcha here which I still think is (also) an * problem is that > > if you set any features on "tTwW" etc it filters the character and it does > > not send it out over PSTN... > > Me too! :) > We have a server that we wanted the ability to transfer outgoing calls > and so included the ,T option with dial. Once we did that we could not > make remote IVR systems recognize dtmf tones. Our trunks are PRI ISDN. > We are using a T410P card connected to a channel bank for analog phones. > (No SIP in this case) Removing the outgoing transfer fixed the IVR > issues. This is definitely an asterisk issue. > > Don Pobanz > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >"Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety." -- Ben Franklin (1759) **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************
Doug Crompton wrote:> Ok well I am not crazy! This seems like such an important issue I am not > sure why it has lasted for so long. DTMF is the backbone of everything we > do here. Without it we would not have calls!! At least get the DTMF stuff > right. I feel a little guilty complaining since this is free but it is > also used in some high level demanding situations (not mine) and thus is > not a trivial issue.In my experience, PSTN DTMF problems are usually a volume issue. Play with the receive and transmit gains on the SIPura FXO port. Of course when doing SIP you need your DTMF mode (RFC2833 is what I recommend) correct before you start trying to fix the PSTN DTMF issues. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery.
----- Doug Crompton <doug@crompton.com> wrote:> I just wish that Digium would have made a statement saying they are > aware > of the problem and are working on it. I did kind of dig that out of > the > Digium archives but it seems to have ended back in March. If theWe have done exactly that. It is being worked on, and Asterisk 1.4 will have a vastly improved RFC-2833 implementation. However, for most people (90% or more), the existing implementation works just fine and they have no complaints. The problem you mentioned regarding pressing single keys that are lead-ins to feature codes is something separate, and may very well be a legitimate bug. I've just reviewed the code in Asterisk 1.2.x and it appears to do the right then when a feature DTMF sequence 'times out' and is not completed (that is, it sends the already-pressed DTMF digits to the other party), but if you can reproduce the problem and provide a complete console trace please feel free to open a bug on bugs.digium.com. -- Kevin P. Fleming Senior Software Engineer Digium, Inc.
> We have done exactly that. It is being worked on, and Asterisk 1.4 willhave a vastly improved RFC-2833 implementation. However, for most people (90% or more), the existing implementation works just fine and they have no complaints. When is 1.4 expected to be released? - Gabe