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, Peder <pe...@networkoblivion.com> wrote:
From: Peder <pe...@networkoblivion.com>
Subject: Re: [asterisk-users] Robotic sound sometimes
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
Date: Saturday, 13 February, 2010, 4:25 AM
There is a statistics area and you can select sip or voip calls to see
calls. It shows packet loss, jitter, latency, out of sequence packets, etc.
It can even play them back, so you can check where the loss is and play back
the call to see if the noise is in the same spot. Here is some info from
the wireshark website:
http://wiki.wireshark.org/VoIP_calls
-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 12, 2010 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Robotic sound sometimes
On Fri, 12 Feb 2010, Peder wrote:
> Since it is sporadic, my guess would be network latency / packet loss
> /jitter to ITSP. You may have lots of capacity and they may claim to
> have lots of capacity, but what about the links between you and them.
> Who knows when/if there is loss and latency and jitter there. Setup
> wireshark to grab some calls and when someone complains about it, look
> at the stats for that call. It will tell you if there is loss or
> latency or jitter.
Do you know of a link that explains how to detect latency or jitter? My
wireshark skills are pretty non-existent. Does it have a wizzy filter that
will tell you or do you need to check the timestamps of the RTP packets?
Any chance you could write up your technique?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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