Displaying 20 results from an estimated 2000 matches similar to: "Max TDM calls per asterisk box"
2010 Jul 16
1
g729 codec loading
Hello Everyone,
I've successfully registered my g729a licenses.
When i try to load the module from asterisk Cli i got the following error
*Error loading module 'codec_g729a.so':
/usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after
reloc: Permission denied*
* loader.c:795 load_resource: Module 'codec_g729a.so' could not be
2011 Apr 20
2
Call files or AMI originate for mass outbound call
Hello Guys,
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
thanks
--
*Adolphe CHER-AIME
Network / VoIP Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-4280*
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2010 May 19
2
Asterisk Cluster
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to have your advice on hardware, software and so on . What i have in
my plan is a cluster of servers with quad PRI cards.
I will appreciate your advice.
Thank you all .
--
2010 May 30
6
How to use one single IP as origination
I have an Asterisk with multiple IP's, on the same subnet. When a call comes
in, I need to send it back out via SIP, but need that only one IP is used as
originating IP for all calls.
For example
machines has
192.168.50.3
192.168.50.4
192.168.50.5
....
but when I originate the second leg of a call, the IP address that is
supposed to be read as source IP must be 192.168.50.5, regardless of how
2010 Jul 16
6
Video IVR Asterisk ?
Hi
Is it possible to receive video calls using Asterisk and then process them
as an IVR ? One of our clients wants to set-up a video IVR system in the US
and we are evaluation possible options.
Also, what is the bandwidth of receiving a video call in US ? What protocols
and codecs are supported and does it work on DID numbers ? Can I rent a
hosted solution for this ?
Thanks in anticipation of
2010 Jun 29
5
What‘s the best operating system suggest for Asterisk 1.6.2.9
hi, list
i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.
i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
Thanks for your help.
--
Thanks for your supporting,
have a nice day.
Sucan
2010 Nov 08
3
Get the Uniqueid of Action Originate in the AMI
Hi to all.
I'm begin a use the AMI and i have the need to get the uniqueid from the
call i have generate using the Action Originate. Anyone can help me?
When I generate these commands:
action: Originate
channel: SIP/101
application: Dial
data: SIP/100,120,Ttr
The only response I get when the call is answered, is this:
Response: Success
Message: Originate successfully queued
Thanks a
2010 Oct 18
5
IAX2 works one direction, but not the other...
2010 Jun 06
1
Assign dhadi channel to several groups
Hello guys,
I was wondering if it's possible to assign a dahdi channel to two
diferent groups.
Thanks
Adolphe Cher-aime
From my Iphone
2010 Sep 08
3
IPSec on asterisk
Hi,
I am trying to configure ipsec on asterisk. Have configured
/etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in
same folder.
Have run racoon. Still I can't receive calls.
Can anyone please tell if any extra step is needed.
Thanks
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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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2012 Nov 03
2
dahdi 2.6.1+2.6.1 compile fails
I am trying to compile a dahdi module from checkout:
svn co http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.6.1+2.6.1
with ubuntu 3.5.0-17-generic and gcc 4.7.2
Error on compile is:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47:\
error: 'NULL' undeclared (first use in this function)
This is identical to the error reported in this patch fix:
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 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 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 Oct 22
5
dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook?
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 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 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