Displaying 20 results from an estimated 600 matches similar to: "How can I eveluate trailing numbers in extensions.conf?"
2005 Aug 03
1
app_dbodbc for asterisk stable 1.09
Hi,
Has anyone manage to comile app_dbodbc or ast_data with the latest
stable release (1.09). If so can you give some guidence on howto do it
as I have trouble getting either working.
Umar
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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2005 Feb 22
2
Zap timing device
Dear list,
I have been using asterisk for some time now. However I have never
used it with any of the digium or compatable cards (Purely used for
SIP).
I understand that for using Meetme, I need to have a timing device,
which could either be hardware or zrdummy etc (I am not using any
right now).
Can someone tell me if the timing device is needed for voicemail and
other applications too?. I am
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.
--
2005 Mar 27
3
Can't get format_mp3 to work for music on hold
Hi Guys,
I am having trouble trying to get format_mp3 working to play music on hold.
I have followed the instructions in the read-me and the wiki however
it seems after un-installing mpg123, asterisk is not even attempting
to play MOH.
My musiconhold.conf is
; Music on hold class definitions
;
[classes]
[moh_files]
default = >/var/lib/asterisk/moh-native
;default =>
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
2004 Apr 13
1
DNID Digits - Australia
Hi,
Yet another question, now that I have callerid working correctly, I'm
trying to work out how to utilise the different numbers I have. I have a
100 number range allocated to my E1/PRI/OnRamp service.
My incoming calls are handled like this:
Advertised/published number is an analogue line terminating on a X101P.
If the analog line is busy, it has a call diversion to the PRI on a
TE405P
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
2005 Mar 08
2
Asterisk Management API
Hi all,
I am trying to write an application to monitor queues using the
Asterisk Management API.
So far I have had some level of sucess, basically reverse engineering
the protocol and the event messages using ethereal etc.
I know there are a couple of pages on the Wiki that attempt (no
dis-respect to who ever did it as it has been a great help) to
document the API and was wondering if there
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
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
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)
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
2004 Dec 01
14
ASTCC configuration problem
hi
Today I?ve installed, apache 2.0.52, mysql-4.1.7, asterisk-perl-0.08
and ASTCC prepaid card aplication from CVS, so now I have access to
the astcc-admin.cgi from web server
http://asterisk/cgi-bin/astcc-admin/astcc-admin.cgi and I?ve been able
to create the database from "Configure" menu but I have some doubts to
continue:
- Do I have to reinstall asterisk with mysql support?
-
2005 Aug 26
1
bridging sip to capi, no playtones back to caller
I've the following setup :
sip phone -> ser (auth and routing) -> asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack
-- Executing Dial("SIP/2.7.184.61-08152880",
2006 Jan 13
3
FastAGI Command Execution
I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands!
Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it
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
2006 Nov 20
4
Auto recording calls?
Howdy, folks.
I'm having a problem finding a way to auto-record calls (both incoming
and outgoing). I know how to make it so either party can initiate
recording, but I want it done as soon as both ends are connected (or
prior to that if that's what it takes). It's probably right in front
of me and I'm just missing it. Any help would be much appreciated.
Thanks,
Jay
2012 Feb 20
3
Park and PARKINGDYNAMIC
I have been trying to get the dynamic parking working.
For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work.