[Asterisk-Users] 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.

> 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.

If you analyze this interaction in terms of real telephony standards,
iax should _not_ be issuing the "ringing" function back to livevoip,
but rather providing an inband audio ringback.

So, your only choice is to live with it, or jump through hopps to
play an audio ringback within your extensions.conf context.