I am having a big problem with DTMF. I have a customer using an Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem is that when they dial into a conference bridge or IVR where they have to enter a code they always get an error. Either some numbers are duplicated or missing. They use Teliax for calls to the USA and Protel in Mexico. Both carriers have the same problem so I think Asterisk could be at fault here. I have tried using dtmfmode as auto, inband, info and rfc2833 but get the same result. I have tried ulaw, alaw and g729 as the codec for the calls but have the same problem. Could this be a timing issue? Since there are no zap channels we use ZTDUMMY for timing. The server usually runs only about 15 simultaneous calls so the load is not heavy on the processor. What is the best method to debug DTMF issues? Do I have to sniff the SIP packets? -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20081003/f68d4bb2/attachment.pgp
Hey Carlos,> What is the best method to debug DTMF issues? Do I have to sniff the > SIP packets?The best method to debug DTMF issues depend on how you receive those DTMF digits. Assuming you can use SIP INFO for the DTMF, that means the DTMF digits are not really DTMF :-), that is, is not audio, with SIP INFO the digits will be received by Asterisk as part of the SIP signaling protocol and therefore you can easily spot them using "sip debug peer myxpeer". If I were you I'd try that and see what I am really receiving before drawing any other conclusion. Moy -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire