I have just setup a small queue implementation for one
of my branch offices, replacing a 16 year old key system
that had a hacked together pseudo call queuing feature.
The 'agents' are not dedicated to the queues and want to
be able to logon and get one call only from the queue.
I know this is odd
This was so interesting I had to move it to its own thread!
Is anyone using this script? How does it perform compared to the older
WonderShaper script?
-M-
==
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-Original Message-
On Fri, Oct 31, 2008 at 11:39:31PM +, Robert Lister wrote:
> On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
> > Hi,
> >
> > I have a strange problem with our Asterisk installation. Outgoing calls
> > are handled by the following lines:
> >
> > exten => _0[2-9]X.,1,Set(CALLE
I think everyone is missing the point of the question. He wants to know
if the user's shell is set to rasterisk, can they then use the CLI to get
a command shell.
The answer is "yes, they can", and in that case it may not be such a
good idea. As someone else suggested, you can run a shell with
Thanks Kristian I will checkout the new script and see how it goes!
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Friday, October 31, 2008 1:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [
Hi,
Many thanks for the info. Just a question, will I need to isntall anything else
now?
Will i only need to install libpri and Asterisk from now on?
Best regards and thanks,
Christian
On 2008-11-01 at 09:06 David Klaverstyn wrote:
>Nothing changes except for the files.
>
>/etc/zaptel.conf beco
On Wed, Oct 29, 2008 at 01:13:56PM -0400, Leah Newmark wrote:
> From time to time, voicemail.conf would go blank. We finally tracked it
> down to happening when someone attempts to change their password.
> It seems the file is touched, but not written to, and we're left with a
> blank voicemail
On Sat, Nov 01, 2008 at 12:38:52AM +0100, Dima wrote:
> Setting the user's shell to /usr/sbin/rasterisk works. On login user
> gets into asterisk CLI if asterisk is running (user just has to have
> write permission to /var/lib/asterisk.*).
How does that user "login"?
--
Tzafrir Co
On Fri, Oct 31, 2008 at 08:18:32AM +0100, Stefan Guenther wrote:
> Hi,
>
> I have a strange problem with our Asterisk installation. Outgoing calls
> are handled by the following lines:
>
> exten => _0[2-9]X.,1,Set(CALLERID(num)=0403${CALLERID(num)})
> exten => _0[2-9]X.,2,SET(CALLERID(num)=$
Setting the user's shell to /usr/sbin/rasterisk works. On login user
gets into asterisk CLI if asterisk is running (user just has to have
write permission to /var/lib/asterisk.*).
> On Fri, Oct 31, 2008 at 11:11:08AM +0100, fadey wrote:
> > Hi, everyone
> >
> > I'm investigating if I could give a
Then you should read the READMEs right now. See the 3 upgrade info
files as well as any other READMEs.
Christian wrote:
> Hello,
> Many thanks for the info.
> OK, I didn't know that. I just installed it. Usually I read the included read
> me files and so on but not at this time.
> But I will be
Nothing changes except for the files.
/etc/zaptel.conf becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf becomes /etc/asterisk/chan_dahdi.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Sent: Saturday, 1 November 2008 8:49 AM
To: [EM
Hi again,
I am using the A400P card.
many thanks,
Christian
On 2008-10-31 at 23:49 Christian wrote:
>Hello,
>Many thanks for the info.
>OK, I didn't know that. I just installed it. Usually I read the included
>read me files and so on but not at this time.
>But I will be able to use my old zaptel
Hello,
Many thanks for the info.
OK, I didn't know that. I just installed it. Usually I read the included read
me files and so on but not at this time.
But I will be able to use my old zaptel hardware that i used with v1.4?
Many thanks,
Christian
On 2008-10-31 at 22:31 Darrick Hartman wrote:
>T
Time for you to discover who's your dahdi...
Asterisk 1.6 used dahdi and not zaptel.
--Original Message--
From: Christian
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk insta
Hi all,
I've just installed the latest v1.6 release of Asterisk. First, I isntalled
libpri.
Then i installed zaptel with make config at the end of the isntallation as I
usually do.
Then I installed Asterisk.
However, there is no zapata.conf file in /etc/asterisk. I isntalled the sample
configura
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then af
On Fri, Oct 31, 2008 at 03:12:44PM -0500, Brent Davidson wrote:
> I ran into almost this exact same problem when I first installed
> asterisk. My company uses a virtualdomain hosted by our isp. We'll
> call it mycompany.com for example. When I first set everything up I
> wasn't able to send a
I ran into almost this exact same problem when I first installed
asterisk. My company uses a virtualdomain hosted by our isp. We'll
call it mycompany.com for example. When I first set everything up I
wasn't able to send any mail from the asterisk server even though it was
on an accepted IP.
We have a situation where it's sometimes taking Asterisk 17-19 minutes to post
CDR's, both over the AMI, and over the MySQL socket. It seems however that they
are logged locally to /var/log/asterisk/cdr-csv/Master.csv right after the call
is terminated.
Anyone got any idea what's causing this?
You can use individual Asterisk boxes to feed a subset of the 3000 phones (ie
96 analog ports) That would be 1 4 port card with 4 T1 channels banks. You
could think of these as RTs (Remote Terminals) and then you can use DunDI to
have the calls 'routed' to the correct RT. The good thing about th
> Date: Fri, 31 Oct 2008 11:39:43 +0200
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk with SC440 Dell(Big Problem)
>
> Hi
>
> On Fri, Oct 31, 2008 at 03:35:23AM +, Edwin Quijada wrote:
>>
>> I have a Dell SC440 with Centos and Asteri
Hi
below are my configs:
pstn(e1)--->asterisk (span1)->legacy pbx(connected via span2)-> legacy
analog extensions.
my dial plan is like callers dial into asterisk(span1) and they are connected
to the agents via the legacy pbx (which is in sync with asterisk on
span2)the prob is when
Hi,
Any one with any experience with VoIP hard phones or adapters
supporting speex? I looked around google but could not find any phones
supporting speex.
raj
___
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asterisk-users mai
Hello All,
I'm having an issue where asterisk doesn't hear any audio after transferring to
voicemail. Here is the dial plan and console output.
DIAL PLAN
[voicepulse-in]
exten => _14259491337,1,NoOp(Incoming call from VoicePulse)
exten => _14259491337,2,Ringing
exten => _14259491337,3,Wait(1)
ex
Hi Paolo,
You can always supply a command with MixMonitor to rename the file after
the call is completed. At that point the variables are evaluated and you
can probably rename/move the recording to which phone answered the call.
It looks like Monitor doesn't have this feature, but I'm sure you co
Hello there,
I appreciate any help about this problem that I can't figure out...
I need to record all my calls: this is pretty easy using Monitor() before
the Dial().
eg:
exten =>
425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb)
exten => 425,n,Dial(${PHONE
Peter Galiovsky wrote:
> Does anyone have any idea what should I try next?
>
Either contact Digium support directly or the people you bought the G729
license from. You're more likely to get the assistance you need in a
shorter period from these people than this list.
__
if your ISDN card was installed new.. try disabling any built in PCI
interfaces.. that's what i see for now.. and let me know if it works.-- AHD
Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944
618286 USA: +1 347 562 2308 > Date: Fri, 31 Oct 2008 22:32:37 +1100> Fro
On Fri, Oct 31, 2008 at 11:11 AM, Hans Witvliet <[EMAIL PROTECTED]> wrote:
> Zombie processes ;8
DeadAGI();
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
2008/10/29 Alex Balashov <[EMAIL PROTECTED]>
> You can use Cisco phones as long as they have a SIP image.
and as long as users are ready to learn its english GUI as you can't
localize GUI using SIP image and Asterisk
>
>
> Kev Szaszvari wrote:
>
> > Hi there
> >
> > Our company is using the Lin
Hello,
I have trouble activating the G.729 codec in Solaris. It seems the
module doesn't try to read the license file at all.
Although the module loads (it's listed in show modules), "show g729"
gives "No such command" error and this is posted in the messages log:
Oct 27 18:09:40 NOTICE[19336] c
On Fri, Oct 31, 2008 at 10:32:37PM +1100, Lee, John (Sydney) wrote:
> I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5
> on DELL PE2950 and using ISDN-10.
What device?
> I thought about cutting over to production tonight when I noticed a
> serious problem.
>
> SIP calls ar
Jim Boykin pisze:
> We are running Asterisk SVN. We are facing a strange and repetable
> problem. All outgoing call gets terminated in approx 20 minutes.
> Asterisk initiates BYE message to the remote end and call terminates.
>
Sesion-timer set but not supported by sip-peers?
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5
on DELL PE2950 and using ISDN-10.
I thought about cutting over to production tonight when I noticed a
serious problem.
SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times
or someone called in a few times,
On Fri, Oct 31, 2008 at 11:11:08AM +0100, fadey wrote:
> Hi, everyone
>
> I'm investigating if I could give asterisk CLI access to one of our
> clients.
> If I add that user to asterisk group and set his shell
> to /usr/sbin/rasterisk, is there a possibility for a user to brake our
> of asterisk C
Hi,
We are running Asterisk SVN. We are facing a strange and repetable
problem. All outgoing call gets terminated in approx 20 minutes.
Asterisk initiates BYE message to the remote end and call terminates.
Can anyone help?
Thanks
Jim
___
-- Bandwidth
Hi!
I think I saw a command "!", which would escape to a shell. But I'm not
sure. Unfortunitely I can't look it up at the moment, because I compiled my
asterisk for full debug. Just enter your CLI and type at the
prompt. I think I only saw this in the latest SVN.
But your client could do
Hi, everyone
I'm investigating if I could give asterisk CLI access to one of our
clients.
If I add that user to asterisk group and set his shell
to /usr/sbin/rasterisk, is there a possibility for a user to brake our
of asterisk CLI to normal shell?
Thanks in advance
___
On Fri, 2008-10-31 at 10:54 +0100, randulo wrote:
> Morning!
>
> This may be the "day of the dead" in some regions, but we expect the
> usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM
> UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This
> Sunday, I believe the
Morning!
This may be the "day of the dead" in some regions, but we expect the
usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM
UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This
Sunday, I believe the USA falls back to Standard time. Future VUC are
still at 12 N
Hi
On Fri, Oct 31, 2008 at 03:35:23AM +, Edwin Quijada wrote:
>
> I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox
> D110PG, T1, when a person calling from the PTSN will listen to them
> but then begins to distort the voice I heard that name.
This symptom is not clea
Hi,
I would like to get musiconhold from a sound card. This is because I
want to kind of be a DJ and easily change the music playing, etc.
However, I followed the instructions at
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and
other tutorials on the net but no success. I have
[
Hi everyone,
I have encounter a problem that when i use the dtmf signalling, i can not
received the callerid from the caller, my asterisk versions is 1.4.21. I
think asterisk have not solve the problem,anyone can solve the problem, help
me , Thanks in advance.
--
Best regards!
jordan pan
Locat
Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> What kernel config do you use?
OK, Just for the Record:
The problem is gone when CONFIG_GROUP_SCHED is disabled!
Regards
Sven
--
"linux is evolution, not intelligent design"
(Linus Torvalds)
/me is [EMAIL PROTECTED], http://sven.gegg.us/ on the Web
Hi,
I have a strange problem with our Asterisk installation. Outgoing calls
are handled by the following lines:
exten => _0[2-9]X.,1,Set(CALLERID(num)=0403${CALLERID(num)})
exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} =
040321]?04030:${CALLERID(num)})})
exten => _
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