[Asterisk-Users] Re: fine-tuning asterisk questions

[M.Hockings]

Thanks William,

Excellent description, I think I understand what needs to be done, now I 
just need to figure out how to best implement it!

I'll dig out the dialplan tonight and try and re-describe problem #2 
with it.

Mike


William Piper wrote:
> 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 
> <mailto: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