Displaying 20 results from an estimated 4000 matches similar to: "CEL Documentation"
2010 Oct 23
2
1.8 Console Welcome Message
With previous Asterisk versions when running asterisk -r a welcome
message is displayed with the version. I just upgraded to 1.8 and
noticed it is not appearing. All I get is Verbosity is at least 3 and
the console prompt. I looked at main/asterisk.c and still see the
welcome message code. Any idea why it is not being shown?
Ryan
2012 Aug 22
3
Asterisk 1.8 and 11
Just a little questions, what's the difference between asterisk 1.8
and asterisk 11?
Best regards.
2011 Apr 25
4
The new ConfBridge application is now in Asterisk Trunk!
Howdy,
I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598
If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely
2010 Sep 22
2
Can't cross compile asterisk 1.6.2.13 on arm using ltib
Hi,
I can cross compile asterisk 1.4.21 on arm (imx27) using ltib
I want to cross compile the new version 1.6.2.13 but there is an error when
I execute the commands :
./configure --build=i686-pc-linux-gnu --host=arm
make menuselect
The configure seems ok, I have the result info :
*configure: Package configured for:
configure: OS type : none
configure: Host CPU : arm
configure:
2007 Jun 26
6
Cisco 7941 localized menus with SIP firmware
Hi,
Has anyone met any success, installing localized (ie non-english) menus
within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco Call
Manager but as they are managed by an Asterisk server, I'm looking for a
workaround.
Any advice ?
Regards
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2010 Nov 17
6
How many Asterisk PBX operating in the World?
Hi,
Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide?
Thanks and hope the community will not reject my curiosity! :)
Best Regards,
Vallu
Sevana Oy
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2010 Oct 20
4
Recommendation for a new server
Hello list,
What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and
a not much busy website, i.e. getting 500-1000 hits a day.
Thanks,
Zeeshan A Zakaria
--
www.ilovetovoip.com
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2010 Oct 14
1
MySQL and Channel Event Logging
Hey all, sorry if this has been covered, but I've not found anything after a
couple hours' worth of googling. I can see (and I'm familiar with) all the
usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any
reference to MySQL and the new CEL logging tool other than ODBC. Is this the
only method available to use MySQL with CEL at this time?
Thanks,
Sherwood
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2010 Oct 21
10
Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
2010 Oct 14
6
Audiocodes firmware
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body text="#000000" bgcolor="#ffffff">
<font size="+1">Does anyone have links to the most recent audiocodes
2010 Jun 24
4
OT: Bandwidth calculations
Hi,
I know some of you are very experienced as to the working of
networks. I wondered whether there is some accepted way of determining
bandwidth needs based on the network traffic over time. For example,
looking at the figures for the network traffic through the server
interface, we have hourly, daily and monthly figures. If everything
were linear, taking the hourly figure and dividing it by
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
2010 Oct 18
5
Same extension registering over eth0 and eth1
Hello list,
I need to know how to deal with a redundant network with only one asterisk
server, which is receiving registrations from the end points on both of its
ethernet ports. This means extension 201 is registering both from eth0 and
from eth1.
Is there a way/software which can act as a middle man between asterisk and
the ethernet ports, and by default sends registrations to asterisk only
2010 Jul 24
4
getting some segmentation faults with 1.8
I downloaded the latest 1.8 (27922) but got some segmentation faults.
The first one was when it loaded cdr_odb, and so I changed menuselect
not to compile that one, but the second one was when it tried to load
chan_agent and so I stopped there to see if anyone else was seeing
this. The agents.conf is all commented out except for [general] .
Anyone know what is happening?
Thanks.
P.S. I deleted
2010 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2010 Sep 14
9
Speech To Text on linux with asterisk
Hi,
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
or any available tool open source for speech to text .
Regards
Dhaval
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2010 Sep 11
8
No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
Hi Everyone,
I have a provider whose DID used to come into the box just fine but recently
stopped working. Nothing has been changed on our end.
Here is what I get when doing "sip set debug peer PROVIDER":
Sending to 123.123.123.123 : 5060 (no NAT)
^^^^ That is ALL I am getting with sip debug turned on.
With Allow Anonymous SIP set to YES, then the call comes in properly and you
see
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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