Thanks! Actually, I want to share the asterisk DB using multiple
asterisks. So I use NFS to share the whole directory
/var/lib/asterisk in order to share files including astdb of asterisk.
However, there is not what I expected. Say, UA1 registers asterisk1
and UA2 registers asterisk2.
Hi Everyone,
I am running Asterisk 1.2.7 Zaptel 1.2.5 on CentOS 4.3 on a Dell
PowerEdge SC420. I was running an older version of Asterisk (can't
remember what, but was using the wcfxs kernel module) under Gentoo
Linux and succsessfully had Asterisk talking to my TDM400P card.
However on my
I need to put an Asterisk server in a remote office where only ADSL is
available. Maximum of 8meg downstream 646k upstream.
Is this an adsl2 line ?
If yes ask your provider if it supports channel bonding. You could use 2
adsl lines as one. All load balancing etc is done at the dslam side.
Hi Mr. Jones,
Mr. Jones wrote :
snip
It seems there are probably two routes, but I'm not sure of the
limitations of each.
1. Shared call appearances. This would seem to be the most similar to
what we currently have where we have stations/DNs for 3 executives on
3 assistants phones. Of course
Hi,
I am working out to establish calls
between two asterisks PBX'esusing H323 channels. I am using SJ phones as
the H323 clients.
The scenario looks like,
SJ phone(444) --- Asterisk PBX 1 --- H323channel ---
Asterisk PBX2 --- SJ phone(555).
I was able to make the calls and channel was
--- [EMAIL PROTECTED] a écrit :
Hello,
Asterisk crash with chan_oh323.so
i use asterisk 1.2.9 asterisk-oh323-0.7.3
What's wrong ?
ACF|192.168.0.11:1720|4762_endp|2505|900:dialedDigits|903:h323_ID=903:dialedDigits=903:h323_ID=903:dialedDigits|false;
Tzafrir Cohen:
On Sat, Jul 22, 2006 at 09:02:48PM +0200, Frank Darner wrote:
Hi,
I have problem to set up an X100P clone card.
Installation of zaptel was successful.
Also modprobe of zaptel, ztdummy and wcfxo without problems.
kernel: wcfxo: module not supported by Novell, setting U
configs look fine to me
it has to be a module problem im guessing
may be, but I dont know whats wrong
Zaptel (1.2.6.) installation was without errors and I did not get an error
when loading the modules
try rmmod for zaptel and wctfxo or w/e then modprobing again(must be root)
if that
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written a patch which
seperates out the the sip registry into a new table. If this is stable
and tested, then you might want to use that with an ARA configuration
and have all your
Why don't you use the syntax that I mentioned in my first reply?
According to http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+localnet
The correct syntax is:
localnet=192.168.0.0/255.255.255.0
Keyword localmask is deprecated in asterisk 1.2... And btw, you should
have seen it in the
This is probably an easy one but i have not been able to fix it.
Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine and every
time i try to make it i get an
Configure: error: termcap support not found
Make: *** [editline/libedit.a] Error 1
Ive installed termcap-compat using
Am Sunday 23 July 2006 06:39 schrieb Tzafrir Cohen:
On Sat, Jul 22, 2006 at 09:02:48PM +0200, Frank Darner wrote:
Hi,
I have problem to set up an X100P clone card.
Installation of zaptel was successful.
Also modprobe of zaptel, ztdummy and wcfxo without problems.
kernel: wcfxo:
It did not work, how can I put in some user intervention so that any numbers they dial will send them to a message? Restrict their outbound calls and a get a message to contact administrator instead of a busy signal.
-- Original message -- From: "brandon kruz" [EMAIL
What is the output from 'cat /proc/zaptel/*'
After delete of all Asterisk files and complete new install I got
something:
# modprobe zaptel modprobe wcfxo
linux:/proc/zaptel # cat /proc/zaptel/*
Span 1: WCFXO/0 Generic Clone Board 1 RED
1 WCFXO/0/0
but # ztcfg -
is
Anybody know about a softphone (open source) that support G726-32 codec ?
Thanks in advance
roberto
--
Ing. Roberto Pereyra
ContenidosOnline
Looking for Linux Virtual Private Servers ? Click here:
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If you are using phones attached to a ZAP FXS port the immediate=yes
will work. Otherwise, some SIP phones (Grandstream for instance) allows
you to enter an autodial number. It depends on what is providing the
dialtone to the handset. If your device does not support autodial, then
the next
Thanks for the response, its looks logical, for some reason the authentication is not working for me, I'm using a Linksys Phone adapter and here is a sample dial plan in extensions.conf and also SIP channels.
exten = 8407,1,Dial(SIP/8407,80,rt) ; permit transferexten = 8407,n,Authenticate(9461)
[9507] is the incoming User ID. user=8407 is the outgoing User ID.
Do you really want them to be different?
Dial() will stop processing of the dialplan until the call ends. Do you
really want this?
r option to Dial will force a ringing sound to the caller, even if the
caller should be
That was a typo its corrected to [8407] but problem still persist with original questions though.
-- Original message -- From: "Eric "ManxPower" Wieling" [EMAIL PROTECTED] "[9507]" is the incoming User ID. "user=8407" is the outgoing User ID. Do you really want them to
On Wed, May 24, 2006 at 08:54:09PM -0400, Matt wrote:
Then bristuff may be the way to go. However, I read this on the wiki
Note: Using bristuff breaks PRI support, so you cant have both bri
and pri in the same server.
That's not good! I need PRI.
I never had this problem, using a BRI and a
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Hello all,
I've been trying to play with asterisk (after a two month break) and am
having some problems getting my SIP connection to a third party provider
to work. In the asterisk console I notice:
- -
debian*CLI set verbose
perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'On 7/23/06, Frank Darner
[EMAIL PROTECTED] wrote:
What is the output from 'cat /proc/zaptel/*' After delete of all Asterisk files and complete new install I got
You could put the phone in a context such as context=restricted in sip.conf
In extensions.conf put a context
[restricted]
exten = _.,1,Answer
exten = _.,2,Authenticate(8675301)
exten = _.,3,Goto(whateverdialcontext,whateverexten,whateverpriority)
replace Allison's recording for authenticate
Tom Lynn:
perhaps not what you're looking for, but reading thru your config, it looks
like you've mis-spelled 'echo cancel' as 'echo cancle'
you are right, typo
thank you
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
You can do it one of two ways:
1) make the SIP device dial a predefined number when the user picks up
the phone. You do this in the SIP device. Check the manual for that
device for detail on how to do this. It's normally called hotline.
In extensions.conf have Asterisk run Authenticate
which is exactly what I said if you read the whole thread :-)
Eric ManxPower Wieling wrote:
You can do it one of two ways:
1) make the SIP device dial a predefined number when the user picks up
the phone. You do this in the SIP device. Check the manual for that
device for detail on how to
- [EMAIL PROTECTED] wrote:
Im trying to install asterisk 1.2.10 on a new debian 3.1r2 machine
and every
time i try to make it i get an
Configure: error: termcap support not found
Make: *** [editline/libedit.a] Error 1
Install the libncurses-dev package.
--
Russell Bryant
Software
Hi,
Thanks to Julian, my internal/external Nat problem is solved.
For anyone else working from outdated example files, the format with
localnet and localmask on separate lines is no longer supported.
The localnet line must also have the netmask included as per Julian's
example below or it will
Hi Friends,We have two internet connections (lines) from two Internet Service Providers in our office. So, we have two public IP addresses. We are using one connection for our LAN and to providing internet to our office staff. We are not using second connection. Now, we have installed
Now, we have installed "Asterisk" and using for International dialing
with Second connection. Now, I have installed "XLite" softphone in our staff
systems. I tried to connect our XLite with our Asterisk server. But, our
XLite softphone is unable to connect with Asterisk server. I have given
look udev rules???On 7/23/06, Frank Darner [EMAIL PROTECTED] wrote:
Tom Lynn: perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'you are right, typothank you___
--Bandwidth
El vie, 21-07-2006 a las 18:53 -0400, Russell Bryant escribió:
On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote:
Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to
bind to 10.152.58.9:5060: Address already in use
It looks like another application on your
Thanks Sebastian -
You're right - I have limited experience in this area :)
I think the idea below is workable, except we actually want it to work
in the other direction - sort of.
Essentially we want the receptionist to screen the calls when she's
available. The executive should have option
Am Sunday 23 July 2006 20:58 schrieb Walter Willis:
look udev rules???
the problem was related that ztcfg did not find zaptel.com
-c /etc/asterisk/zaptel.conf has solved this issue
#ztcfg --help -c filename -- Use filename instead
of /etc/zaptel.conf
my failure, I should read man page
-enable|6002|OUT) in new stack
-- Executing GotoIf(Zap/2-1, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Zap/2-1, recordingcheck|20060723-185730|
1153695447.0) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck
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RR wrote:
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written a patch which
seperates out the the sip registry into a new table. If this is stable
Save you searching:
On Fri July 21 2006 18:33, Woodoo People .pGa!
[EMAIL PROTECTED] wrote:
don't forget the following:
if canreinvite=yes, asterisk will NOT stay in mediapath, so, it going to
ask both parties to negotiate codec, and say hello to the stream. (if
both parties supports g729, and can negotiate it,
I have been scouring the net the last couple of days looking for some
kind of tutorial or walkthrough on setting up a h.323 channel in asterisk.
What I need to do is basically this:
I have a client who wants to be able to connect to me via h.323 and make
a local phone call (local to me, he is
Hi everyone,
I am using Asterisk on CentOS 4.3 with a TDM400P and have managed to
get things up and running except this one part.
My /etc/sysconfig/zaptel configuration has only one MODULES directive
enabled MODULES=$MODULES wctdm
However when I start asterisk it loads the wct1xxp module.
Do you mean the patch can use to replace asterisk DB by ARA?
On 7/24/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:
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RR wrote:
Unplug, I'm sure there are other people with better ideas but if you
see on sineapps, I remember someone having written
i have been wondering about how the useragents work since a month or two. i have tried every document possible...could not find the answer.
if anyone could tell me about the useragents, how they work, what are the factors that are considered while choosing a UA, what makes a particular UA best,
Gents,
does anyone have a conformation about meetme working well in with ARA?
I found this particular fix put in somewhere around jan'06
http://bugs.digium.com/view.php?id=5702
Sounds interesting, but not clear from the status if this is actually
been merged in newer releases or safe to apply
I posted about this some while back, and at that point was told the
remote end is broken, nothing we can do about it.
The problem: for whatever reason, some CallerID names come in broken.
There is an example CLI trace shown below.
My question: is there anything I can do to fix this, since
Hi Franck,
NOACPI and the sound must be more clear.
And, of course, have you tell to /usr/src/zaptel/zconfig.h and
/usr/src/asterisk/Makefil what kind of processor you have and enabled MMX if
possible before to compile ?
Good Luck !
Francois BERGERET,
France.
-Message d'origine-
De :
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