Displaying 12 results from an estimated 12 matches for "drishti".
2009 Aug 01
1
how to setup incoming calls not to use authentication
Dear all,
In Sip.conf file how to setup incoming calls not to use
authentication?
Please provide some steps to do it..
Thanks...
Regards,
Velusamy
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2009 Oct 30
2
asterisk 1.6 enable cdr_mysql
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
"module show" is showing "cdr_addon_mysql.so"
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file to enable mysql support?
Comping cdr_mysql.conf from previous installation does not do anything, calls aren't recorded.
--
2010 Mar 11
2
Codec preference
How can I set the prefered codec between 2 calling parties ??
My Grandstream supports G729, alaw and gsm... in this order.
The Zoiper softphone has alaw and gsm as codecs... in that order.
Although there should be a matching codec found, my Grandstream can not
call the Zoiper softphone.
CLI shows :
[Mar 11 17:47:21] WARNING[22367]: channel.c:3340
ast_channel_make_compatible: No path to
2009 Dec 18
2
To Asterisk AMI Gurus - Tacking issue with originate
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
2010 May 26
1
Getting "ghost" transfer or music on hold
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="Tahoma">Hi Everybody,<br>
<br>
I´m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ...
In some calls, i get an atxfer or musiconhold in the middle of
2009 Jun 30
0
Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE>
...-007f0e98",
"3600") in new stack
[Jun 26 22:24:19] DEBUG[3670]: pbx.c:3179 pbx_extension_helper: Launching
'Wait'
-- Executing [calllegwait at from-manager-core:1] Wait("SIP/902-007fe948",
"3600") in new stack
--
Regards,
Prince Singh
W: http://www.drishti-soft.com
B: http://blog.drishti-soft.com
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2005 Mar 13
2
How can I eveluate trailing numbers in extensions.conf?
Checkout
http://www.voip-info.org/wiki-Asterisk+variables
I believe that should have the answer for you.
furthermore assuming that your number is always going to be 12 digits.
exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number.
Hope this helps.
Umar
On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote:
> Hi,
>
> this
2010 Mar 23
4
Safe_asterisk doesn't exists???
Hello my friends,
I'm very worry about a problem i'm having...my asterisk got freez some
times, every 5 or 6 days with NO trace in /var/log/asterisk/messages
What i want to know is if safe_asterisk has something to be with this?
This is what i have on my server:
[root at mypbx ~]# ps -A | grep asterisk
9118 ? 00:01:30 asterisk
[root at dreampbx ~]# ps aux | grep asterisk
root
2009 Dec 30
1
problem with ring being sent to caller
I am using asterisk 1.6.0 and -- not all the time -- when a caller comes
in and my ivrdials an extension, the ring he gets sounds like a modem
handshake instead of the normal ring tone and it only sounds once even
if the phone is not picked up. Anyone seeing this -- the logs look fine
as far as I can tell.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you
2010 Apr 13
1
Interesting One Way Audio
I have an Asterisk box, 1.4.30 with a PRI.
A Mitel 3300 is connected to the Asterisk box via SIP trunking.
When a user calls from the Mitel through the Asterisk box the user can speak
but can not hear the far end.
But - when I route the Mitel user to echo() it works, send and receive. The
Mitel user also can record and playback greetings.
One thing I have noticed is that when the Mitel user
2009 Dec 28
2
Registering with a static peer?
I've been using a couple of Polycom 501 phones in my home Asterisk setup. I set
up each phone in sip.conf to be static, i.e. host=<phone ip address> so that
registration wasn't required. This has worked fine for me for a couple of years.
Now I just bought a Polycom 335. Since the 501's are now obsolete, I had to go
through the steps required in order to have separate
2004 Aug 13
11
asterisk in india
Does anyone know if the E1 cards that digium sells work in India. Also are
there any distributers for those cards in India. By E1 cards I mean E100P,
TE410P or TE405P
--
regards
Vikram (http://www.vicramresearch.com)