<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;"><DIV>hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan.</DIV>
<DIV> </DIV>
<DIV>sip.conf</DIV>
<DIV> </DIV>
<DIV>[general]<BR>port = 5060<BR>bindaddr = 0.0.0.0<BR>context = others</DIV>
<DIV>[2000]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2001]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2002]<BR>type=friend<BR>context=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2003]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV> </DIV>
<DIV>[2004]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2005]<BR>type=friend<BR>contex=outside<BR>secret=1234<BR>host=dynamic</DIV>
<DIV> </DIV>
<DIV>[2006]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2007]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV>[2008]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2009]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>[2010]<BR>type=friend<BR>contex=internal<BR>secret=1234<BR>host=dynamic</DIV>
<DIV><BR>############################################################################################</DIV>
<DIV>vi /etc/asterisk/extensions.conf</DIV>
<DIV>[from-zaptel]<BR>exten => s,1,wait(2)<BR>exten => s,n,dial(sip/2000)<BR>exten => s,n,dial(sip/2001)<BR>exten => s,n,Playback(tt-weasels)</DIV>
<DIV>[others]<BR>include => internal<BR>include => outside</DIV>
<DIV>[inside]<BR>exten => _20XX,1,Dial(SIP/${EXTEN})<BR>exten => _20XX,n,VoiceMail(<A href="mailto:${EXTEN}@others,u">${EXTEN}@others,u</A>)<BR>exten => _20XX,n,Hangup()</DIV>
<DIV>[outside]<BR>exten => 2001,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2001,n,Hangup</DIV>
<DIV>exten => 2002,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2002,n,Hangup</DIV>
<DIV>exten => 2003,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2003,n,Hangup</DIV>
<DIV>exten => 2004,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2004,n,Hangup</DIV>
<DIV>exten => 2005,1,Dial(Zap/1-1/${EXTEN})<BR>exten => 2005,n,Hangup</DIV>
<DIV><BR>this is the log when i am calling from exten 2000 to outside</DIV>
<DIV><BR>Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)<BR>Verbosity is at least 3<BR>[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call from '2002' to extension '919369613616' <STRONG>rejected because extension not found</STRONG>.</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>any help n support will be highly appreciated<BR><BR>--- On <B>Sat, 13/2/10, Jonathan Addleman <I><jono@redowl.ca></I></B> wrote:<BR></DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"><BR>From: Jonathan Addleman <jono@redowl.ca><BR>Subject: Re: [asterisk-users] problems with 1.6<BR>To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><BR>Date: Saturday, 13 February, 2010, 4:54 AM<BR><BR>
<DIV class=plainMail>I'm still unable to do much with my new 1.6 installation. I just tried<BR>reinstalling, and using the standard debian configuration files, with<BR>just the necessary modifications, in case I had some legacy stuff in<BR>there from earlier versions that was interfering. I'm testing in a xen<BR>domU with debian's asterisk package, version 1:1.6.2.0-1<BR><BR>When I try to connect a channel into a conference, I still get all sorts<BR>of "Unable to write to alert pipe" and then "Exceptionally long voice<BR>queue length queuing" errors. I've tried this with meetme and<BR>appkonference, with no difference in the error messages.<BR><BR>This happens with various sorts of channels - audio playing, eagi<BR>scripts, sip connections. For example, I just tried this through the<BR>manager interface:<BR><BR>Action: originate<BR>Channel: Local/conference@veco/n<BR>Context: veco<BR>Exten: playaudiofile<BR>Priority: 1<BR>Variable: tour=test<BR>Variable:
dir=<BR>Variable: conference=ConferenceA<BR>Variable: provider=teliax<BR>Variable:<BR>extravalue=/var/www/vecotourism/media//transcoded/audio/long-asterisk<BR>Variable: title=Long test sound (long-asterisk.wav)<BR><BR><BR>with this in extensions.conf:<BR>exten => playaudiofile,1,Answer<BR>exten => playaudiofile,n,Wait(1)<BR>exten => playaudiofile,n,Playback(${extravalue})<BR><BR>exten => meetme,1,Answer()<BR>exten => meetme,n,Wait(1)<BR>exten => meetme,n,MeetMe(${conference}_${tour},1qd)<BR><BR>exten => conference,1,Answer()<BR>exten => conference,n,Noop(Trying to start conference ${conference}_${tour})<BR>exten => conference,n,konference(${conference}_${tour})<BR><BR><BR><BR>Seems this should be very straightforward, but it isn't working. What<BR>might be wrong?<BR>-- <BR>Jon-o Addleman - <A href="http://www.redowl.ca/" target=_blank>http://www.redowl.ca</A><BR><BR>--
<BR>_____________________________________________________________________<BR>-- Bandwidth and Colocation Provided by <A href="http://www.api-digital.com/" target=_blank>http://www.api-digital.com</A> --<BR><BR>asterisk-users mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-users" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></DIV></BLOCKQUOTE></td></tr></table><br>
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