[Asterisk-Users] fine-tuning asterisk questions

[William Piper]

For Problem #1:
exten => _X.,1,SetGroup(${EXTEN})
exten => _X.,2,GotoIf($[${GROUPCOUNT} = 1]?104:3)
exten => _X.,3,Dial,SIP/username
exten => _X.,104,voicemail(u${EXTEN})
exten => _X.,105,hangup
This will limit the amount of incoming calls to "1" and send everything else
to the VM.

For Problem #2:
I'm not sure what you are asking. Perhaps post your dialplan for this
problem & we will take a look.

bp

On 6/4/06, M.Hockings <veeshooter at hockings.net> wrote:
>
> I have asterisk running more or less ok but I would like to turn off
> call waiting and be selective about the incoming sip connections.  This
> is running asterisk 1.2.8 with a fxs and fxo card and a configured voip
> (sip) line.  Currently I'm using freePBX 2.1.1 to configure asterisk.
>
> Problem 1) if someone is on the phone already and another call comes in
> for an already engaged extension I want it to go to voicemail directly
> rather than have that distracting call-waiting beep going on.
> As far as I can tell I have turned off call waiting in the zaptel config
> files.  What else should be set to avoid call-waiting ?
>
> Problem 2) Incoming sip calls from my voip provider get rejected unless
> I allow anyone to connect with sip. I have an incoming route set up with
> the right DID that matches the DID that asterisk picks out but it still
> rejects the call.  Any suggestions about how to get this to work without
> allowing any sip connection?
>
>
> Mike
>
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