<div>
Thanks all, I will follow the recommendation of moving my sip extensions to a db or use perl to check if the extension exists. I run tests with dialstatus, hangupcase, chanisavail and the return codes are the same if the extension doesn't exist on sip.conf or if the extension is not registered.<br><br>Regards<br>Richard<br>
<blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">----- Original Message -----<br>
From: "Eric Wieling" <eric@fnords.org><br>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br>
Subject: Re: [asterisk-users] Calls to sip extensions not defined<br>
Date: Sun, 23 Mar 2008 07:50:21 -0500<br>
<br>
<br>
exten => i is for IVRs.<br>
<br>
You would need a wildcard catchall extension like the one below.<br>
Unfortunately you are doing the classic newbie mistake of thinking you<br>
can have a simple dialplan by making the SIP user/account ID be the same<br>
as the extension. Eventually you will realize this is a bad idea.<br>
<br>
If you think you have to use wildcards (you don't) then you would want<br>
to check the value of DIALSTATUS or HANGUPCAUSE after the Dial, before<br>
the Voicemail and handle the exception using a Goto.<br>
<br>
[default]<br>
exten = _1[1-3],1, dial(sip/${EXTEN},10)<br>
exten = _1[1-3],n, voicemail(${EXTEN}@default,u)<br>
exten = _1[1-3],n, hangup<br>
<br>
exten = _XXX,1, playback(pbx-invalid)<br>
exten = _XXX,n, hangup<br>
<br>
<br>
Steve Edwards wrote:<br>
> On Sat, 22 Mar 2008, R. B. wrote:<br>
><br>
>> [default]<br>
>> exten = _1[1-3],1, dial(sip/${EXTEN},10)<br>
>> exten = _1[1-3],n, voicemail(${EXTEN}@default,u)<br>
>> exten = _1[1-3],n, hangup<br>
>> exten = i,1, playback(pbx-invalid)<br>
>> exten = i,n, hangup<br>
<br>
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