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* <[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>> 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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