[Asterisk-Users] Re: Problems with incoming calls

[Joel Jn-Francois]

> > 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.