<div>For Problem #1:</div>
<div>exten => _X.,1,SetGroup(${EXTEN})<br>exten => _X.,2,GotoIf($[${GROUPCOUNT} = 1]?104:3)<br>exten => _X.,3,Dial,SIP/username<br>exten => _X.,104,voicemail(u${EXTEN})<br>exten => _X.,105,hangup</div>
<div>This will limit the amount of incoming calls to "1" and send everything else to the VM.</div>
<div> </div>
<div>For Problem #2:</div>
<div>I'm not sure what you are asking. Perhaps post your dialplan for this problem & we will take a look.</div>
<div> </div>
<div>bp</div>
<div> </div>
<div><span class="gmail_quote">On 6/4/06, <b class="gmail_sendername">M.Hockings</b> <<a href="mailto:veeshooter@hockings.net">veeshooter@hockings.net</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I have asterisk running more or less ok but I would like to turn off<br>call waiting and be selective about the incoming sip connections. This
<br>is running asterisk 1.2.8 with a fxs and fxo card and a configured voip<br>(sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk.<br><br>Problem 1) if someone is on the phone already and another call comes in
<br>for an already engaged extension I want it to go to voicemail directly<br>rather than have that distracting call-waiting beep going on.<br>As far as I can tell I have turned off call waiting in the zaptel config<br>files. What else should be set to avoid call-waiting ?
<br><br>Problem 2) Incoming sip calls from my voip provider get rejected unless<br>I allow anyone to connect with sip. I have an incoming route set up with<br>the right DID that matches the DID that asterisk picks out but it still
<br>rejects the call. Any suggestions about how to get this to work without<br>allowing any sip connection?<br><br><br>Mike<br></blockquote></div>