i need some information on Lucent EXS 2000 class four switch. Can some one
give me the URL for the web site where i can find some info on it.
Thanks
Regards
Ayaz Gul Aga
Network Manager
2B-Technologies
- Original Message -
From: Mark McKibbin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
I try to use oh323 package from inaccessnetworks for asterisk, but after
make and make install that package, I have this WARNING message hwen a try
to launch asterisk from shell command line...asterisk -vvvc...
[liboh323wrap.so]WARNING[1024]:
Larry Creech wrote:
Trying to configure voicemail with H.323 all I get is the following errors
when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it
might be a simple fix.
Are all these aliases present in [register] section in oh323.conf?
Is there a context associated with
O -fPIC-c -o chan_agent.o chan_agent.c
chan_agent.c: In function `login_exec':
chan_agent.c:595: parse error before '' token
chan_agent.c:602: parse error before '' token
make[1]: *** [chan_agent.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
chan_agent.c: In function `login_exec':
chan_agent.c:595: parse error before '' token
chan_agent.c:602: parse error before '' token
make[1]: *** [chan_agent.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
My fault...
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Tom De Wispelaere wrote:
2. run some cronscript every 5 seconds that checks the database and checks
the calls to be made.
cron only wakes up once a minute...
Cheers,
Holger von Ameln
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I have recorded my name for my voicemail, but my name is not played in
the Directory application.
I am using Voicemail2 (if that matters). Could this be a context issue?
Do I need to change any
of the configuration files?
-Derek
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Mark,
That is exactly what happened. I commented out the music on hold,
none of my Customer Service reps like it. They would rather listen to
silence.
:)
Not to be a pest, but is there any updates on the Data fix you were
working on?
John
On Wednesday, June 18, 2003, at 08:46 AM, Mark
Hi,
I have * running on a SuSE Email Server II, which uses an older
libc-version. Calls to res_ninit et al. from enum.c prevent asterisk
from building on that system. Could that possibly be changed to use
res_init and res_search in order to provide backward compatibility with
these systems ?
hi
-modem.conf :--
msn=240862922
incomingmsn=240866365,6365
device = /dev/ttyI2
group=1
device = /dev/ttyI1 ; ttyI3, ttyI4
-extensions.conf ;---
[sip]
exten = _XX,1,Dial,Modem/g1:BYEXTENSION
(Sjphpone) Call to : 024076
result :
--Executing
hi,
i installed asterisk and works very well, the only problem is that
when i try to call a direct number of a company that has a normal PBX
i got this error:
to 10.8.210.153:5060
== Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro)
-- Executing Goto(SIP/a.sampietro-f7be,
I have found out why I am having problems with the directory
application.
The problem is that the Directory application is not searching through
contexts properly.
Example:
extensions.conf
[default]
exten=*,1,Directory(company)
exten=556,1,Directory(default)
exten=555,1,Voicemail2,u555
...
[I'm reposting this to the asterisk-users list, since it seems to be a
bit more active.]
Hello,
I started messing with Asterisk few days ago, so my overall knoledge
about it is still fairy superficial.
I think I found an issue with MP3Player; it can be reproducted with this
extension:
exten =
You can call setmusiconhold app and as an argument call class silence,
off, or whatever non-existant class and it works now.
Martin
On Wed, 18 Jun 2003, TC wrote:
Yea, I have faked that with a silent mp3,
but to do it right it should also be a config flag in the agent.conf file
for each
- Original Message -
From: Hervé THIBAUD [EMAIL PROTECTED]
To: Asterisk-Users [EMAIL PROTECTED]
Sent: Wednesday, June 18, 2003 5:05 PM
Subject: [Asterisk-Users] ISDN BRI
result :
--Executing Dial(Sip/roseau-6163,Modem/g1:BYEXTENION) in new stack
-- Called g1:024076
--
On Wed, 2003-06-18 at 11:38, Angelo Sampietro wrote:
hi,
i installed asterisk and works very well, the only problem is that
when i try to call a direct number of a company that has a normal PBX
i got this error:
to 10.8.210.153:5060
== Accepting call on 'SIP/a.sampietro-f7be'
On Wed, 2003-06-18 at 11:11, Chip G wrote:
Thanks to this group, I am making great progress in
getting up and running with Asterisk! I actually
connected with a SIP client last night and made a
phone call (still getting some error reports from
Asterisk -vvv during the process and
On Wed, 2003-06-18 at 10:27, Holger von Ameln wrote:
Hi,
I have * running on a SuSE Email Server II, which uses an older
libc-version. Calls to res_ninit et al. from enum.c prevent asterisk
from building on that system. Could that possibly be changed to use
res_init and res_search in
Does anyone know if this was implemented? If not then where should I look to
try and make the mod?
Thanks
Tan
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, April 28, 2003 9:03 AM
Subject: Re: [Asterisk-Users] SNOM 200 and MWI??
Hi Mark,
Steven Critchfield wrote:
While it may be possible, I won't comment on the technical side of that
fix.
Linux distros are free, and upgrading to a new distro is probably
preferred so you don't risk any exploits on that machine. If you are
using it in production for some other task, you may wish
sae for grandstream ?
On Wednesday 18 Jun 2003 7:55 pm, Test wrote:
Does anyone know if this was implemented? If not then where should I look
to try and make the mod?
Thanks
Tan
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, April
On Wed, 18 Jun 2003, Test wrote:
Does anyone know if this was implemented? If not then where should I look to
try and make the mod?
MWI works fine on the Snom 100 and 200. On the 200, the amber message
waiting light will flash whenever a voice message is left. It is much more
visible than the
I just picked up a couple CAC Access Bank 1s loaded with FXS that should be
arriving shortly. Does anyone have one that they use with Asterisk? If
so, would you be willing to shoot me a note with your current configs? I'm
not very familiar with CAC/etc, and it would save me countless hours of
Is it possible to specify a 'wrap-up' time in a queue so agents will
have a specified amount of time to complete tasks between calls unless
they hit a key on the phone? As it is they can recieve a call moments
after they hang up with no 'down time'. Thanks
Jim Friedeck
Thanks to both Iain Oliver, but it looks like the i4l driver wasn't my
problem. I had been testing using a SIP softphone (X-Lite) and that appears
to be the problem. See the
[Asterisk-Users] soft phones -- voice quality tuning
thread for exactly the same problem posted by another user. After I
Hello Tielman Koekemoer
E1 is used in the world except for North America and one or two other places.
It consists of 30 speech or data channels and 2 signalling (1 for framed signalling,
and one for channel signalling)
E1 is superior to the North American T1 system over here it's called
i hope this helps, because this is similar (and yes, i cvs'd the
day 'after' the supposed cvs commit for the fix.
Martin Pycko wrote:
Describe that a little bit.
The call came on what interface and then you dialed what interface
and how did you park it ? You pressed a flash button or '#' key
geez i'm tired, bt's would help more eh G
attached..
Richard Lyman wrote:
i hope this helps, because this is similar (and yes, i cvs'd the
day 'after' the supposed cvs commit for the fix.
Martin Pycko wrote:
Describe that a little bit.
The call came on what interface and then you
On Wed, 18 Jun 2003, Bradley Greep wrote:
Hello Tielman Koekemoer
E1 is used in the world except for North America and one or two other places.
It consists of 30 speech or data channels and 2 signalling (1 for framed signalling,
and one for channel signalling)
E1 is superior to the
A bit of a me too, I'm also getting dropouts calling from Xten Lite no
matter what protocol I use. However using Windows Messenger 4.7 the sound is
clear.
I did notice that when X-Lite (logs/registers) with asterisk that the
following warning is displayed on the asterisk console.
On Wed, 2003-06-18 at 17:01, Christopher Arnold wrote:
Hi all,
i would likte to do an asterisk -rx show channels regulary from cron and
save the results. Easy it may seem, but when running the command from cron
sip*CLI is the only result.
Even expect seems to have problems with getting
Hi all,
i would likte to do an asterisk -rx show channels regulary from cron and
save the results. Easy it may seem, but when running the command from cron
sip*CLI is the only result.
Even expect seems to have problems with getting results from asterisk.
Is it possible to solve the above
On Wed, 2003-06-18 at 17:18, Alex Zarubin wrote:
Hello,
Question for developers: what is the asterisk way to integrate with
ASR (speech recognition)?
Question to the community: has someone done anything in this
direction?
On the first glance that shouldn't be too hard. One part is
About 2 weeks ago, I made the first announcement to the list about the first IAXCLIENT cross-platform builds.
What we had then, was a command-line application which could make a single outgoing call.
We've since done quite a bit of work on the clients, and the client now has:
1) A working
Why not write a [insert favorite scripting language here] script to use
the Asterisk manager interface?
I've wrote quite a few little perl scripts each doing their own specific
function. Works quite well.
Jeremy
Christopher Arnold wrote:
Hi all,
i would likte to do an asterisk -rx show
Is it possible that you are using an old version of the Xten software? We
had problems with GSM in the earlier builds but that was rectified several
weeks ago. Could you tell me what build you are using?
Also, there is a new version of Xten software coming out tomorrow morning
which enhances QOS
On Wed, 18 Jun 2003, Jeremy McNamara wrote:
Why not write a [insert favorite scripting language here] script to use
the Asterisk manager interface?
I've wrote quite a few little perl scripts each doing their own specific
function. Works quite well.
Could you give us a pointer to info
Hi Eric
looks like you are with Xten ...
Is there any chance that Xten would put IAX inside the xten lite client ???
Is it possible that you are using an old version of the Xten software? We
had problems with GSM in the earlier builds but that was rectified several
weeks ago. Could you tell me
This works for me.
Martin
#!/usr/bin/perl -w
use Socket;
use IO::Handle;
socket(SOCK, AF_INET, SOCK_STREAM, getprotobyname('tcp'))
or die Cannot create a socket: $!\n;
connect(SOCK, sockaddr_in(5038, inet_aton('localhost')))
or die Cannot connect to the manager port\n;
There isn't much to set... Just use B8ZS and ESF, clock source line or
internal, just be sure to use the opposite on the * box.
Also, you know you'll need a T1 crossover cable right? If you need to know
the pins, let me know, or just search the list, its been posted.
I would go ahead and
I have an extension setup with voicemail, for incoming calls on an X100P
card. It quite often will record about 15 seconds of dialtone... I'm
guessing that it picks up the line after the outgoing line has been
disconnected.
Has anybody else run into this problem? Shouldn't chan_zap be detecting
Anyone in New Zealand using AsteriskPBX? If
so, what hardware are you using to connection to Telecom's lines?
archives ?
On Thu, 19 Jun 2003 17:07:35 +1200, Aaron Martin wrote:
Anyone in New Zealand using AsteriskPBX? If so, what hardware are you using to
connection to Telecom's lines?
.
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