[Asterisk-Users] Re: Problems with incoming calls

[Rich Adamson]

> > > 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.

Maybe its livevoip switch dependent (guess on my part). My incoming
calls via 217.160.244.186 are just fine.

> > > 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.

In all likelihood, sixtel is honoring the iax "ringing" command (since
they are obviously using asterisk). Pure guess on my part is the livevoip
has an older or heavily modified version of asterisk, but really
don't have a clue.

If you sniff the iax packets to both, you'll see the iax control pkt
going to both; one vender's reaction is different then the others. That's
the best we can say/observe without more detail from those vendors.