Try playing with these; exten => s,n,DigitTimeout(3) ; Set Digit Timeout to 3 seconds exten => s,n,ResponseTimeout(20) ; Set Response Timeout to 20 seconds
Many Regards. Danny Froberg On Thursday 24 March 2005 09.24, Joel Jn-Francois wrote: > > > 1) When an incoming call to my DID number is initiated, a prompt is > > > > played so that the caller > >can enter an extension number or > > > > > zero for the operator. However, at least 30%-50% of the time the > > > > digits that are entered from > >the touch tone phone is slightly > > > > > different from what is received by asterisk. There is usually double > > > > digits when only one of > >those digits were entered. For > > > > > example I would enter 4071, but asterisk would receive 4007 or 4077 > > > etc. > > > >I'm not having the above problem at all; works fine. If you have a dtmf > >statement in your incoming iax.conf context, remove it. > > That was the first thing I looked for when I started having that > problem. I do NOT have any DTMF statements in my IAX, SIP or Extension > configuration files in asterisk. I have gone through all the configuration > files and have not found anything that may contribute to this > problem. However, how would you explain that the fact callers never > experience that problem with Sixtel DID numbers. The only difference > between Livevoip and sixtel DID that I am using is that I am getting 1800 > DIDs from Livevoip and with Sixtel I am using local DIDs for my area. > > > > 2) If the extension number was correctly received by asterisk and I > > > > pass the call to a SIP > >extension I would then lose Audio > > > > > until the phone is answered. If I simply pass the call to a SIP > > > > Extension without playing any > >prompts and I don't use the answer > > > > > command before I transfer the call, then I can hear the ringing audio > > > > just fine. > > > >This is a known issue with livevoip.com service. It's my opinion this > >is really a design issue within asterisk, but Mark disagrees. > > > >The problem is * must answer the incoming iax call from livevoip in > >order to execute the IVR menues. When the caller then dials an extension > >number, * responds to livevoip with "ringing" expecting livevoip to > >provide the ringing to the caller. Since the call is in "answered" > >mode, livevoip is simply ignoring the iax "ringing" command. Its my > >opinion the livevoip is properly ignoring that iax function as the > >call path has already been cut through, end-point to end-point. > > I under what you are saying perfectly. What I don't understand is why I do > NOT have that problem with other providers like Sixtel. Do you think that > Sixtel responds back providing the ringing to the caller? Is it possible > for Sixtel to know that the call was not really answered but was > transferred to an extension. I have no idea what Sixtel is doing, but > maybe Livevoip should look into a way around this issue. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
