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