<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'>
i did and its not working here is console output. We have 8910-8920 meetme conf room. below i am dialing 8991 for test invalid and its not working.. <br><br><br>Packet timed out after 32000ms with no response<br> == Using SIP RTP CoS mark 5<br> -- Executing [7580@from-sip:1] Goto("SIP/7527-00000030", "ivr-meetme,s,1") in new stack<br> -- Goto (ivr-meetme,s,1)<br> -- Executing [s@ivr-meetme:1] Answer("SIP/7527-00000030", "") in new stack<br> -- Executing [s@ivr-meetme:2] Wait("SIP/7527-00000030", "1") in new stack<br> -- Executing [s@ivr-meetme:3] BackGround("SIP/7527-00000030", "conf-getconfno") in new stack<br> -- <SIP/7527-00000030> Playing 'conf-getconfno.ulaw' (language 'en')<br> -- Executing [s@ivr-meetme:4] WaitExten("SIP/7527-00000030", "20") in new stack<br> == CDR updated on SIP/7527-00000030<br> -- Executing [8991@ivr-meetme:1] MeetMe("SIP/7527-00000030", "8991,cMp") in new stack<br> == Parsing '/etc/asterisk/meetme.conf': == Found<br> == Spawn extension (ivr-meetme, 8991, 1) exited non-zero on 'SIP/7527-00000030'<br>shirley*CLI><br><br><br><br><br>> Date: Wed, 6 Apr 2011 14:37:20 -0700<br>> From: asterisk.org@sedwards.com<br>> To: asterisk-users@lists.digium.com<br>> Subject: Re: [asterisk-users] asterisk meetme invalid extension<br>> <br>> On Wed, 6 Apr 2011, satish patel wrote:<br>> <br>> > I have following dialplan for meetme and i want if someone type wrong <br>> > meetme extension it should say invalid extension. But look like <br>> > following doesn't work. its just hangup if i type wrong number. how to <br>> > fix this code..<br>> > <br>> > exten => i,n,Playback(pbx-invalid)<br>> <br>> The priority should be 1.<br>> <br>> -- <br>> Thanks in advance,<br>> -------------------------------------------------------------------------<br>> Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST<br>> Newline Fax: +1-760-731-3000<br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br> </body>
</html>