Tony Mountifield wrote: > a separate context [foo], with a wildcard extension _X., which will match > any extension of two or more digits. I then put the extension number into > the parameter list for the AGI. > > So instead of generating "data: test.agi|12345" in the call file you > generate "extension: 12345" and that then finds its way through to the > AGI command. > > Sorry, I missed that. Well, as long as what we wanted to test is tested.... > OK, this solution is technically the same as mine, concerning channel > handling, so if you are still not getting DTMF, that is a problem. > > Please try adding a delay parameter to Answer, or a separate Wait line: > > exten => _X.,1,Answer(0.5) > > or: > > exten => _X.,1,Answer > exten => _X.,n,Wait(0.5) > > It might be that something in the channel is not finished setting up > before you call your AGI. I always have a small delay after answering. > > I did this. I waited as long as 2 seconds. Still the same problem unfortunately. I can see the DTMF in the IAX trace. The AGI trace just sits there... >> So: >> >> ============================== Conclusions ================================ >> IAX Phone => Dial Plan => AGI script >> "Works with DTFM" >> Call File => IAX Phone + AGI script >> "Fails, not DTMF communication" >> Call File => IAX Phone + Extension in dial plan => AGI Script >> "Fails, not DTMF communication" >> SIP => Work always >> ============================== Conclusions ================================ >> >> Unfortunately I am heading out for a week long Europe trip on Monday. >> I'll try to play with this a bit more on Sunday and see if I can make >> some progress. >> > > OK, hope you have some success. > > Cheers > Tony > I have a few other things to try, but that is more like a work-around or at least a different way to attack the problem. Richard Lyman gave me this link http://dynx.net/ASTERISK/gnudialer/agiIVR.agi and I will look into that as well.
Generally I have had quite some issues since upgrading to 1.4 (IVR DTMF fails to be detected sometimes on incoming calls, with SIP => IAX providers I get dropped incoming audio). I am attacking one problem at a time though. // Jonas _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
