Quote
func_odbc can do whatever queries you give it. SELECT/UPDATE are
simply the simplest cases that make it easy to understand the functionality
*OK - but are the Limited to SINGLE STATEMETS or can you have a Muli-Statemnt
Transaction ?*?
Tilghman Lesher wrote:
On Monday 28 April 2008
As this thread discusses, this is a complex subject. Since Dean is
unlikely to blow his own horn, I want to say that I know he will do a
great job hosting the call today, so please try to be there if you
can. You will be able to download the recorded archive in mp3 format
if you can't make the
Hello,
I am sorry to drop in with maybe a dump question. I got my Xorcomm bank
(8 fxs) and I try to get it working on ubuntu without success. I was
looking for a step by step instruction which I could not find. I found
the insert the CD and now it works manual but thats not what I am
looking for.
Hello,
Is there some way to play different files to caller and callee when using
the L-switch in the dial command? I can only send the same sound files to
them with the available options, but it would be nice to have a workaround
with this functionality.
On a sidenote, Digium developers tells me
On Fri, May 9, 2008 at 5:19 AM, Al Baker [EMAIL PROTECTED] wrote:
I think his connect/disconnect is going to take far longer than his 3
queries.
The fact that Asterisk doesn't support sustained MySQL connection from
the DialPlan
is in fact quite a big deal that Digium seems to have its
Hi,
I'm using a Queue in asterisk with IAX2 peers and my agents are also
doing outbond calls.
Actually we are using a GROUP() function like that to prevent users from
beeing Dialed while
in communication:
exten = s,1,gotoif($[${GROUP_COUNT([EMAIL PROTECTED])}=0]?:busy)
exten = s,n,Set([EMAIL
--- Vieri [EMAIL PROTECTED] wrote:
Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?
I guess there's no easy (open-source) solution to this
problem, at least for now (* 1.6?).
I believe Yate2 has something on this but it's still
alpha.
--- Vieri [EMAIL PROTECTED] wrote:
Is there a way of coherently setting up a clustered
queue?
Does anyone have examples/workarounds/links?
I guess this isn't easy to implement, at least in
current Asterisk versions (* 1.6?).
I think Yate2 may have support for clustered queues
but it's still
On Fri, May 09, 2008 at 09:38:04AM +0200, Loic DIDELOT wrote:
Hello,
I am sorry to drop in with maybe a dump question. I got my Xorcomm bank
(8 fxs) and I try to get it working on ubuntu without success. I was
looking for a step by step instruction which I could not find. I found
the insert
The call is still going to show up as the codec with which the voice
segment was established.
Have you viewed the SIP debug messages and confirmed that T.38 is not
being used?
FWIW the device that is receiving the T.38 fax (generally callee)
should be issuing the T.38 re-invite, so you might
9 maj 2008 kl. 03.52 skrev Thermal Wetland:
On Wed, May 7, 2008 at 7:45 PM, Paul Hales [EMAIL PROTECTED]
wrote:
Where are you located?
We are located on the west coast. The person could work remotely,
and we would pay (I should have said that in the first email!)
The west coast of
In menuconfig did you select the g729 music on hold?
Or you can try this one:
http://app5.netjdn.com/~joako/sounds/SampleAudioSource.g729.wav
(remove the .wav extension)
On Thu, May 8, 2008 at 5:46 PM, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello All,
Recently, I build three Asterisk 1.4 box
same is the case in 1.6, i cant set the variable still.
On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple perl script to connect with ast manager api. the
script
tries to set a
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
same is the case in 1.6, i cant set the variable still.
My guess would be that you have a problem with line endings.
All lines received from the manager interface are terminated with \r\n,
not just \n. If you only strip the
Thanx a lot.that was it. will never forget to remove the new
character again. Now its working fine.
On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
same is the case in 1.6, i cant set the
On 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple perl script to connect with ast manager api. the
script
tries to set a channel variable. It extracts the channel name from the
events it recieves
Hello,
You can cluster queues across several servers with VICIDIAL. We have
clients with hundreds of seats taking in hundreds of lines across
multiple Asterisk servers, and the calls are distributed to agents on
all systems.
MATT---
On 5/9/08, Vieri [EMAIL PROTECTED] wrote:
--- Vieri
Al Baker wrote:
I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
Yep, unbelievable.
This is the reason most PBXs are ground start, is there Zap
RE Kushner List Account wrote:
Al Baker wrote:
I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
Yep, unbelievable.
This is the reason most
Matt,
Is there any module or code that would allow this functionality
without using VICIDIAL? I have been able to have about four hundred
agents on a single box, that is not a problem (ULAW SIP only, no TDM).
For distributing queues, I just use the queue timeout value set to a
low threshold.
Just a quick reminder this call is starting in 3 hours from now at 12pm
(that's 12pm est usa or 4pm gmt time).
From the confirmations I've received from people in the asterisk
community there is a real possibility this may come off so if you have
an interest in this space and want to
Probably for the best - you'd look mighty silly otherwise.
(spoken by someone who's done his own share of jumping around, yellng
YES!)
Steve Totaro wrote:
I never kick myself on issues like this. I enjoy the challenge and
the eventual success by jumping around and yelling YES, YES, YES!
I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover,
Hello,
VICIDIAL is completely separate of Asterisk queues and does not use
them at all. It uses a database to keep track of all calls and agent
availability and when a call comes into the inbound AGI script the
system looks for the next agent in line no matter what server they are
on and will
Drew Gibson wrote:
I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
On Friday 09 May 2008 01:39:53 Al Baker wrote:
Quote
func_odbc can do whatever queries you give it. SELECT/UPDATE are
simply the simplest cases that make it easy to understand the
functionality
*OK - but are the Limited to SINGLE STATEMETS or can you have a
Muli-Statemnt Transaction ?*?
Eric Wieling wrote:
Drew Gibson wrote:
I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement
Can anybody help in parsing the manager events efficiently? Any ideas?
On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns
[EMAIL PROTECTED] wrote:
On 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple
Hi,
I verified that this function is really broken since 1.4.17. I tested it in
1.4.19.1, too.
I didn't find information on bug tracker yet. Did someone already add this
there?
Thanks
Paulo
On Thu, Jan 3, 2008 at 4:39 PM, Brent Torrenga [EMAIL PROTECTED] wrote:
Upgraded to 1.4.17 and
Matt Florell schrieb:
The Asterisk Queues code is not really written to handle calls in this
way
I'd say it is not at all written to handle clustered queues. :)
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms.
Sounds a bit scary. Doing everything in
What's the status of ZRTP supported by Asterisk? There was some
discussion on the -dev list and -users list, but it was inconclusive. At
about the same timeframe, a bug (#0010024) was opened and updated for
several months, but has been suspended since late 2007.
Does any version
Benoit Plessis wrote:
So i'm wondering if someone already as made a dialplan function that
could toggle the 'Use' flag of
an agent ? or if this kind of function would be integrated into the core
if i build it ?
This is a slightly different approach, but have you seen the state interface
I would love to be able to issues the necessary Mysql commands to have
true TRANSACTIONS
Such as - Begin Transaction
Select @var=agent.id, agent.exstension where
agent.status='free'
Update agent.status='BUSY' where [EMAIL PROTECTED]
End Transaction
In article [EMAIL PROTECTED],
Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
The Asterisk Queues code is not really written to handle calls in this
way
I'd say it is not at all written to handle clustered queues. :)
I built VICIDIAL around AGIs, manager interface
Russell Bryant a écrit :
Benoit Plessis wrote:
So i'm wondering if someone already as made a dialplan function that
could toggle the 'Use' flag of
an agent ? or if this kind of function would be integrated into the core
if i build it ?
This is a slightly different approach, but
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms.
Sounds a bit scary. Doing everything in MeetMe rooms just doesn't
feel right IMO.
the ability to cluster many Asterisk
Matthew Rubenstein wrote:
What's the status of ZRTP supported by Asterisk? There was some
discussion on the -dev list and -users list, but it was inconclusive. At
about the same timeframe, a bug (#0010024) was opened and updated for
several months, but has been suspended since late 2007.
Benoit Plessis a écrit :
Russell Bryant a écrit :
Alternatively, if you would like to control the usability of an agent
through
the dialplan, then you could use the DEVICE_STATE() function to
create a custom
device state. Then, you could list your custom device as what
app_queue
Benoit Plessis wrote:
This is a slightly different approach, but have you seen the state interface
code that is in Asterisk 1.6? There is a backport of the code for 1.4
floating
around somewhere, I think. It allows you to specify a different device for a
queue member that app_queue will
Al Baker wrote:
I would love to be able to issues the necessary Mysql commands to have
true TRANSACTIONS
Such as - Begin Transaction
Select @var=agent.id, agent.exstension where
agent.status='free'
Update agent.status='BUSY' where [EMAIL PROTECTED]
Russell Bryant wrote:
This is a slightly different approach, but have you seen the
state interface
code that is in Asterisk 1.6? There is a backport of the
code for 1.4 floating
around somewhere, I think. It allows you to specify a
different device for a
queue member that app_queue
On Fri, May 9, 2008 at 10:25 AM, Matt Florell [EMAIL PROTECTED] wrote:
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms.
Sounds a bit scary. Doing everything in MeetMe rooms just
Drew Gibson wrote:
Eric Wieling wrote:
Drew Gibson wrote:
I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a
Thank you for your very kind offer.
After repeatedly re-opening the ticket I finally got a clear specific
answer.
Strangely, in the 30 mins it took for me to take the answer, try it, and
report back the results
they had closed the ticket again so I couldn't report whether their
solution fixed
(Sorry for top-posting. Was just easier this time.)
I'm running Snom 300s with software version 7.1.30. I don't think there
is a whole lot different between them and the other phones apart from
the number of buttons. The Park Orbit thing was really sort of tricky.
There's a setting that
On Thu, May 8, 2008 at 3:59 AM, Russell Bryant [EMAIL PROTECTED] wrote:
Ex Vito wrote:
Now, how to move on to acheive some kind of fault tolerance ?
According to the docs we've studied, DUNDi does not like loops
(which we assume one can limit with low enough TTLs).
Which documentation
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
Thanks
___
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asterisk-users mailing list
To UNSUBSCRIBE
Steve Totaro schrieb:
PS, a really cool app hosted at a telco is totally awesome (except the
price). Check out www.ucn.com
They have APIs and can connect to virtually any database. A call
center in the cloud, brilliant!
All the page says is:
---cut---
ucn.com
revisiting urban territory
On 5/9/08, Steve Totaro [EMAIL PROTECTED] wrote:
On Fri, May 9, 2008 at 10:25 AM, Matt Florell [EMAIL PROTECTED] wrote:
On 5/9/08, Philipp Kempgen [EMAIL PROTECTED] wrote:
Matt Florell schrieb:
I built VICIDIAL around AGIs, manager interface daemons and
agents in meetme rooms.
On Fri, May 9, 2008 at 11:19 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Steve Totaro schrieb:
PS, a really cool app hosted at a telco is totally awesome (except the
price). Check out www.ucn.com
They have APIs and can connect to virtually any database. A call
center in the cloud,
I have been using FC6 for the past 1 year without any problem.
Regards,
Sanjay Rajdev
- Original Message -
From: equis software [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 9, 2008 8:49:23 PM GMT
Hello.
I'm using asterisk in alarm reception system.
The system is DTMF intensive and works well while
all concurrent channels are online. But when one
channel goes hangup the other channels lose tones
while one second.
___
-- Bandwidth and Colocation
equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what
do you think about to use Ubuntu or another distibution??
Thanks
I have run Asterisk on several Fedora versions, Debian, Unslung (on the
NSLU2 or Slug) and recently Ubuntu. My most critical servers
equis software schrieb:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
I prefer Debian, but if everything works well and if you're
familiar with Gentoo why change?
Grüße,
Philipp Kempgen
--
Asterisk-Tag.org
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme. Symptoms include a flood of RTP packets from the
Asterisk server to
On Fri, May 9, 2008 at 11:46 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
equis software schrieb:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
I prefer Debian, but if everything works well and if you're
On Fri, May 9, 2008 at 11:50 AM, Drew Gibson [EMAIL PROTECTED] wrote:
equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what
do you think about to use Ubuntu or another distibution??
Thanks
I have run Asterisk on several Fedora versions, Debian, Unslung
this often becomes a religious discussion.
my free advice worth all you paid for it - Redhat or one of the other
distros that has been Certified on your choice of hardware and which has
a Support Contract on it. Despite what others will tell you... Its a
lonely place when your box no-workie
Hi Aby,
aby azid wrote:
Could anyone explain to me what is this Warning mean and how can I
overcome this
*[May 9 12:53:39] WARNING[3626]: codec_zap.c:155 zap_framein: G.729B
CNG frame received but is not supported; dropping.
[May 9 12:53:39] WARNING[3626]: translate.c:211 framein:
Jason Dixon schrieb:
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme. Symptoms include a flood of RTP packets from
On May 9, 2008, at 12:27 PM, Philipp Kempgen wrote:
Jason Dixon schrieb:
We have a remote office that's having problems with their Polycom.
Sometime after they start a conference, the audio will halt and the
Polycom will become unresponsive. The only recourse is to kill the
Polycom meetme.
David Van Ginneken wrote:
Al Baker wrote:
I would love to be able to issues the necessary Mysql commands to have
true TRANSACTIONS
Such as - Begin Transaction
Select @var=agent.id, agent.exstension where
agent.status='free'
Update agent.status='BUSY'
Hi.
I probed more tests and I detect when a channel goes on-hook or
goes off-hook in the other active channels I listen a short
noise or distortion.
I attempt to select internal clock source from TE121 but with the
same results.
Thanks
Pepe Aracil escribió:
Hello.
I'm using asterisk in
Rizwan Hisham wrote:
Well database really is a bottleneck for me. I am currently trying to do
rating stuff in agi using perl. What im doing is i lookup the rate of
every dialed code for every call from the mysql database using the
longest match technique. The longest match technique costs
On Fri, May 9, 2008 at 11:59 AM, Al Baker [EMAIL PROTECTED] wrote:
this often becomes a religious discussion.
my free advice worth all you paid for it - Redhat or one of the other
distros that has been Certified on your choice of hardware and which has
a Support Contract on it. Despite what
I use Slackware.
On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
Thanks
___
-- Bandwidth and
On Thu, 2008-05-08 at 10:51 -0500, Russell Bryant wrote:
Have you taken a look at the ENUMQUERY() and ENUMRESULT() functions that are a
part of Asterisk 1.6?
The ENUMQUERY() function lets you do a single enum query
From a single zone it seems. So that means a for zone in $ZONES type
of
On Fri, 2008-05-09 at 12:19 -0300, equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what
do you think about to use Ubuntu or another distibution??
if it ain't broken don't fixit,
I used debian, now ubuntu,
...but if your thing needs to be production like,
Hi, I allways use Gentoo y my Asterisk servers and work well, but
what do
you think about to use Ubuntu or another distibution??
I don't know what's prompting you to leave Gentoo but it's gotten much
better with respect to asterisk very recently. That's all I use and I
have to say that after
On Fri, May 9, 2008 at 2:00 PM, David Nedved [EMAIL PROTECTED] wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but
what do
you think about to use Ubuntu or another distibution??
I don't know what's prompting you to leave Gentoo but it's gotten much
better with respect to
On Fri, 9 May 2008, Al Baker wrote:
this often becomes a religious discussion.
my free advice worth all you paid for it - Redhat or one of the other
distros that has been Certified on your choice of hardware and which has
a Support Contract on it. Despite what others will tell you... Its a
I think it's all personal preference I'd never recommend anyone
use ubuntu for anything, honestly.
SLES is my #1 pick with CentOS / PNAELV being a close second...
problem with Cent is there's not central administration like there is
in SuSE (YaST2... it's so simple! gotta setup a network no
Oh, and FWIW a Cisco uses PNAELV as the basis for one of it's most
popular voice products.
http://www.bouncethem.com/5455
On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use
On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED]
wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
Thanks
We use Ubuntu Server on a few of our servers and it's been working fine. We
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Matthew Gibson wrote:
| On Fri, May 9, 2008 at 11:19 AM, equis software
[EMAIL PROTECTED]
| wrote:
|
| Hi, I allways use Gentoo y my Asterisk servers and work well,
but what do
|
| We use Ubuntu Server on a few of our servers and it's been working
On Friday 09 May 2008 10:19:23 am equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what do
you think about to use Ubuntu or another distibution??
Thanks
I have used Fedora 7 8 on both i386 x86_64. I have used the RPMs from
atrpms.net in the past and
Anyone have shared lines (sla.conf) working with Polycom phones? Also,
has anyone figured out if its possible to do 1 button call park with
the softkeys?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
I think there are quite a few aspects to the issue. I agree I've used
the X101p cards which really are a Windmodem with a resistor removed
and I had nothing but echo problems but then again I could have tried
harder.
1) It was an early digium product. I think the Sangoma cards and the
newer
I always do Debian, but, as others have pointed out, six one, half-dozen
the other. I always build from source and typically rebuild the kernel
as well in a lot of cases (i.e. to make ztdummy work well requires a
1000 Hz timing resolution), so it's not really an issue either way.
Linux is
Sherwood McGowan wrote:
David Van Ginneken wrote:
Al Baker wrote:
I would love to be able to issues the necessary Mysql commands to have
true TRANSACTIONS
Such as - Begin Transaction
Select @var=agent.id, agent.exstension where
agent.status='free'
On Fri, May 9, 2008 at 5:32 PM, Florian Hackenberger
[EMAIL PROTECTED] wrote:
Hi!
We are using a red-fone bridge (connected to an E1 line from the Telekom
Austria) to provide TDMoE connectivity to our asterisk server (ubuntu
7.10, asterisk 1.4.10 from ubuntu, libpri 1.4.0-2, zaptel 1.4.9.2
On Fri, May 09, 2008 at 10:49:02AM -0400, Watkins, Bradley wrote:
Russell Bryant wrote:
One problem with that cunning plan is that using custom device states
doesn't work. The code for handling device state changes in app_queue
is looking for a forward-slash in the device name, and
Russell Bryant a écrit :
I don't see why this wouldn't help. You just list the IAX2 peer as the device
Asterisk uses to determine the state of the agent.
Well i've read elsewhere that only SIP peers did support the use flag ?
--
Benoit Plessis +33 6 77 42
On Friday 09 May 2008 16:04:58 David Van Ginneken wrote:
Out of pure curiosity I tried it today using func_odbc on a test server
here. Short version: It worked with some warnings, which could be just
local to my server.
snip
OS error 84 is Invalid or incomplete multibyte or wide character, so
Hi,
I just tried to upgrade from Zaptel 1.4.6 to 1.4.10.1 but it looks like
the Makefile has no rules for the HPEC.
make[3]: *** No rule to make target `hpec/hpec_zaptel.h', needed by
`/usr/src/zaptel/kernel/zaptel-base.o'. Stop.
Can somebody confirm this?
Thanks,
Andres
On Fri, May 09, 2008 at 11:50:59AM -0400, Drew Gibson wrote:
equis software wrote:
Hi, I allways use Gentoo y my Asterisk servers and work well, but what
do you think about to use Ubuntu or another distibution??
Thanks
I have run Asterisk on several Fedora versions, Debian, Unslung
On Fri, May 09, 2008 at 04:55:47PM -0400, Alex Balashov wrote:
I always do Debian, but, as others have pointed out, six one, half-dozen
the other. I always build from source and typically rebuild the kernel
as well in a lot of cases (i.e. to make ztdummy work well requires a
1000 Hz timing
On Fri, May 09, 2008 at 06:36:36PM -0400, Andres wrote:
Hi,
I just tried to upgrade from Zaptel 1.4.6 to 1.4.10.1 but it looks like
the Makefile has no rules for the HPEC.
make[3]: *** No rule to make target `hpec/hpec_zaptel.h', needed by
`/usr/src/zaptel/kernel/zaptel-base.o'. Stop.
A couple of years ago I started my Asterisk carrier with selling
x100p cards and I think I sold around 100 of them in total to people
who could actually contact me and new who I was. Yes, it is a poor
man solution but at least it is a solution. And for the poor man it
is the only thing
Hello list,
I have found some strange problem with the ztdummy timing, maybe you
have already have this problem before, I would appreciate some hints
here or maybe I need to file a bug.
First of all, some background:
I decided to upgrade my testing machine to the current version of
Asterisk
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