asterisk users - Jun 2006 - fine-tuning asterisk questions

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

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@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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>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.
Hey.  I was under the impression
that Asterisk would, by default, send calls to priority n + 101 if the called
station was busy.  Is this not the case?  Why would you have to set up something
special for this to work?

Undrhil

Yes you are correct... by default asterisk will send the call to priority
N+101... what is your point?

You asked about turning off "call waiting".  In the example that I
provided,
if the amount of active calls is "1" then it will forward to VM
without
dialing the exten. That is what you asked for... right?

bp

On 5 Jun 2006 02:59:21 -0000, undrhil.1528785@bloglines.com <
undrhil.1528785@bloglines.com> wrote:
>
> Hey.  I was under the impression
> that Asterisk would, by default, send calls to priority n + 101 if the
> called
> station was busy.  Is this not the case?  Why would you have to set up
> something
> special for this to work?
>
> Undrhil
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>Yes you are correct... by default asterisk will >send the call to
priority
> N+101... what is your point?
> 
> You asked about turning off "call waiting".
 In the example that I provided,
> if the amount of active calls is "1" then
it will forward to VM without
> dialing the exten. That is what you asked
for... right?
> 
> bp
Nope.  I am a different poster just wanting to
clarify (for myself) that Asterisk would do exactly what the original poster
wanted without any special programming.  I wasn't aware that there would be
any kind of notification to the station being called that there was a second
call incoming.  Everything I've read so far just says that if the station
is in use, the call is routed to priority n + 101 as a busy call.

Undrhil

----- William Piper <william.piper@gmail.com> wrote:
> My apologies, I didn't realize I was speaking to someone else.
> As far as I know the dialplan does not need to have the "j"
option to
> do N+101. I'm using 1.2.7.1 without the "j" option and it
jumps fine.
This is true in Asterisk 1.2.x, as the default in the code is to enable jumping
(but the default in the sample extensions.conf file is to have jumping turned
off). In Asterisk 1.4 the default in the code will be to have jumping disabled,
and it will need to be turned on globally (or on an application basis) to use be
used. In Asterisk 1.6 it will be gone forever :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.

----- William Piper <william.piper@gmail.com> wrote:
> By "gone forever" in 1.6... do you mean that even the
"j" in the dial
> plan won't work either? Will it just go to the next priority in the
> event of a congested or busy signal?
That is correct. All the 'j' options will go away, in favor of
channel-variable result codes returned by the applications.
> I assume "goto" will still work... right?
Uhh... yeah. That would be silly to remove it :-)

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.