Setting call-limit=1 in sip.conf will limit the number of incomming (and outgoing?) channels on your SIP device to the number you specifiy (1 in this case). If you want to allow more outgoing, but only 1 incomming, you could do that with some GROUP() checking. Problem is that when there isn't an available channel, Asterisk will return CHANUNAVIABLE or something like that. It's not very helpful. GROUP() checking will allow you to provide a more informative answer to the person who's calling.
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<div><span class="gmail_quote">On 9/21/07, <b class="gmail_sendername">Vieri</b> <<a href="mailto:rentorbuy@yahoo.com">rentorbuy@yahoo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi,<br><br>I would like to know if the following is possible:<br><br>* how to accept only one call at a time on a
<br>particular SIP extension (softphone). I'm referring to<br>incoming calls. Can it be done on the server side or<br>just on the client? ie. all other incoming calls will<br>just be dropped while the extension is busy. In other
<br>words I would like to simulate having just one phone<br>line available. I tried using call-limit=1 in<br>sip.conf. Is this the right way?<br><br>* how to accept only one incoming call at a time for a<br>whole group? If there's an active call on any one of
<br>the extensions, drop the other incoming calls.<br><br><br><br><br><br>____________________________________________________________________________________<br>Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
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