[Asterisk-Users] Re: Problems with incoming calls

[Danny Froberg]

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.