2008/4/25 Matt Watson [EMAIL PROTECTED]:
I haven;t used any BRI cards but... call me crazy but wouldn;t they still
be using Zaptel (even your sangoma... the script might just be configuring
it for you)...
and btw, software echo cancel happens in the zaptel kernel driver...
I think (but I'm
2008/4/24 Andres [EMAIL PROTECTED]:
snip
You can chose 2/4/6 ports to buy and if you need more
just add remoras up to 24 ports.
Is this still usable within 1U server, when you cannot stack PCI cards
like this
xxx
xxx
but you must align them like this
xxx xxx
Hello ppl,
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback of gsm files.
So scouring the internet gave me the solution of installing ztdummy and loading
it as a module.
Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and
2008/4/24 Patrick [EMAIL PROTECTED]:
Hi,
I need to setup an Asterisk box with 4x ISDN BRI links. Looking at the
specs of various cards I favor the Digium B410P and Sangoma A502D
because of hardware echo cancellation. Does anyone have any experience
with either card, good or bad? Which one
Thanks to your answers, but i found more beautiful way to do this -
there is some system variable __TRANSFER_CONTEXT, which defines context
to handle the transfered number, so you can create a new context and
there you can do anything with transfered call - i just hang it up.
It's
Hello,
I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I
want to play the limit file to both caller and callee at the same time, but
it plays the limit file first to the caller and then to the callee. I
searched the list and found someone with the same problem back in
Hi all
Sorry for reposting this, but there haven't been an answer to my
previous message.
I have a minor inconvenience here.
I want to use the m switch in the dial command on our outgoing lines
to play music to the caller whilst asterisk and our telecoms provider
connects the call. It works,
Hi all
I have a small problem here.
We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the
zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under
Ubuntu server.
We have 4 analog line coming into the box via a TDM 800 wildcard with
echo cancel module and quad fxo
Hi List!
I got this error while upgrading zaptel:
make -C firmware hotplug-install DESTDIR=
make[1]: Entering directory `/usr/src/zaptel-1.4.7.1/firmware'
Attempting to download zaptel-fw-oct6114-064-1.05.01.tar.gz
--10:53:09--
Hi,
I've a probleme since few weeks that I don't be able to solve.
I use Thomson ST2030 phone and I've an error when I want to do an
attended transfer with the soft key.
The receiver of the transfer return an : Got SIP response 400 Bad
Request back from 192.168.2.13
The direct transfer with
Hi Guys,
I have client with a Cisco 2690 call manager solution that wants to upgrade
but cannot stomach the costs of continuing with Cisco
The installation will go up to 100 users
The client currently has about 40 Cisco phones and would like to continue
with these phones with the odd Polycom
I'm
On Thu, 2008-04-24 at 19:50 -0400, Matt Watson wrote:
I haven;t used any BRI cards but... call me crazy but wouldn;t they
still be using Zaptel (even your sangoma... the script might just be
configuring it for you)...
Last time I installed them the Sangoma drivers sorta run on top of
zaptel.
On Thu, 2008-04-24 at 17:04 -0400, Andres wrote:
We have tested both and they work fine. The Sangoma is much easier to
install as it does not depend on any other driver, you just run
'setup-sangoma' and follow the instructions. You don't have to fiddle
with the linux kernel or zaptel or
Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT)
From: Steve Edwards [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Forking in Dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
Hi Femi
We have about 50 Cisco 7960s on one site off Asterisk 1.4.18
Its all SIP and it doesn't stress a P3 system much at all.
I am not sure what phones you are using - the 7960s are not hard to configure,
a bit of process to convert from the Cisco Skinny to
SIP (using SIP v8.6) but
Thanks for the answers.
I need to say that this command is executed from another machine, with the
command ssh
because in ocalhost is all ok, with sudo or with root.
I will try that trace to see if it helps me, but the bg probem is start the
service from another machine with ssh .
Best
Does anybody know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
differe and how these different servers will interact as a single
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No probs with the Snoms. No errors show on the * box
when the
On Thu, 2008-04-24 at 20:17 -0400, Steve Totaro wrote:
The Sangoma kernel drivers are different than Zaptel, while running
the install script you are asked if you would like to generate the
Zaptel configs but it is not required, you must also run wancfg to
configure the cards beyond the
On Fri, 2008-04-25 at 08:13 +0200, Olivier wrote:
2008/4/25 Matt Watson [EMAIL PROTECTED]:
I haven;t used any BRI cards but... call me crazy but wouldn;t
they still be using Zaptel (even your sangoma... the script
might just be configuring it for you)...
On Fri, 2008-04-25 at 08:21 +0200, Olivier wrote:
2008/4/24 Patrick [EMAIL PROTECTED]:
Hi,
I need to setup an Asterisk box with 4x ISDN BRI links.
Looking at the
specs of various cards I favor the Digium B410P and Sangoma
A502D
Hmmm,
IMHO this is a fundamental SIP architecture issue.
To meet my understanding of distribution, this would required a proxy
function before the call answer() on the Asterisk. If , in an
ideal world, this proxy function were to be in the path before
answer(), the proxy would need added
A good while back when installing 1.2 there were major issues with UK
callerid. Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information. Sometimes before the first ring or
just after. After applying a third party patch we got it to work. We
were
On Fri, Apr 25, 2008 at 7:03 AM, gmail [EMAIL PROTECTED] wrote:
Does anybody know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
On Fri, Apr 25, 2008 at 01:37:15PM +0200, Patrick wrote:
On Thu, 2008-04-24 at 20:17 -0400, Steve Totaro wrote:
The Sangoma kernel drivers are different than Zaptel, while running
the install script you are asked if you would like to generate the
Zaptel configs but it is not required, you
On Thu, Apr 24, 2008 at 02:04:43PM -0300, Vinícius Fontes wrote:
I have a box running a TE410P with echo cancelling and it works like
a charm. Set up once, forget about it.
That card is E1/J1/T1 (like the Sangoma A10x cards), and not BRI.
--
Tzafrir Cohen
icq#16849755
Are my messages getting through?
This is urgent!! Any pointers?
Benjamin Jacob [EMAIL PROTECTED] wrote: Date: Thu, 24 Apr 2008 23:23:08 -0700
(PDT)
From: Benjamin Jacob [EMAIL PROTECTED]
Subject: Playback / Background / Read choppy, but musiconhold fine, even with
ztdummy
To:
Oops, seems like I didn't realized something: the queue size can't be zero. I
solved the problem by setting maxlen=1 and defining a timeout on the Queue()
app. That way when all the agents are busy, the call gets diverted after
[TIMEOUT] seconds, which is ok to me.
Att
Vinícius Fontes
(Not a real answer to your qustion, but still)
On Fri, Apr 25, 2008 at 10:06:08AM +0200, Ian wrote:
Hi all
I have a small problem here.
We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the
zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under
Ubuntu
2008/4/24 Ken Williams [EMAIL PROTECTED]:
Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with others,
but for some reason when they would
One other thing,
On Fri, Apr 25, 2008 at 10:06:08AM +0200, Ian wrote:
Hi all
I have a small problem here.
We are running Asterisk 1.4.18 and libpri 1.4.3 (how can I check the
zaptel version again?) on an Intel core 2 duo 1.6Ghz, 2GB ram under
Ubuntu server.
We have 4 analog line
Try removing the quotes from the Caller*ID info.
Steve Davies wrote:
2008/4/24 Ken Williams [EMAIL PROTECTED]:
Came upon a problem today that I thought I'd see if it's by design, if I'm
missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server,
In article [EMAIL PROTECTED],
Benjamin Jacob [EMAIL PROTECTED] wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution of installing ztdummy and
loading it as a module.
Did it (using
Actually, the code below works perfectly to fix the transfer disconnect
problem. I was asking of other, better ways, aside from manually
defining on all incoming calls a dummy CID.
To answer Steve's question, using a single TDM400 card for the incoming
PSTN (it's one line, a remote office that
Matt Watson wrote:
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be
using Zaptel (even your sangoma... the script might just be configuring it for
you)...
All Sangoma Asterisk cards **except** the BRI cards will indeed use
Zaptel. But the BRI cards use a
Tony Mountifield wrote:
2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable
the line internal_timing=yes. That should make it play out based on
One other thing comes to mind, make sure you compile with 'Don't
optimize' if you're using gcc 4.2.2
Doug
--
Ben
Congrats on going forward with the project Moises. MFC/R2 support on
chan_zap sounds great, looking forward on trying it out.
Regards,
Lex
On Thu, Apr 24, 2008 at 3:03 PM, Moises Silva [EMAIL PROTECTED] wrote:
Unicall MFC/R2 is activelly maintained. by Moy. Actually it's a backport
of the
Olivier wrote:
2008/4/24 Andres [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
snip
You can chose 2/4/6 ports to buy and if you need more
just add remoras up to 24 ports.
Is this still usable within 1U server, when you cannot stack PCI
cards like this
xxx
On Thu, 2008-04-24 at 18:57 -0700, ronald ramos wrote:
ami using GotoIf correctly?
No, you're not using GotoIf() correctly. In fact, there are several
problems with your syntax. The first problem I spot is that the
GotoIf() application requires an expression, and you haven't supplied an
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution of installing ztdummy and
loading it as
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],60,D(22622#${PIN}#))
Digium specifically has asked me not to use the name Asterisk Users
Conference, but that is mostly what we talk about. We expect that to
change soon, however as we expand into more genera VoIP talk.
Today, FRIDAY April 25 2008 at 12
On Fri, Apr 25, 2008 at 09:53:16AM -0400, Andres wrote:
All Sangoma Asterisk cards **except** the BRI cards will indeed use
Zaptel. But the BRI cards use a totally independent driver that
communicates with the WOOMERA channel. It does not use Zaptel at all
and therefore no software echo
On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis [EMAIL PROTECTED] wrote:
Every CPU core shows up as a separate CPU under Linux. For those that have
hyperthreaded processors, a single core processor will show up as two
processors - assuming you have hyperthreading enabled.
That is interesting.
Been using the Snom 360 and 190 for a while and decided to try the Cisco
7960. The problem I'm seeing is the call terminates between 2:34 and
3:00 minutes. This only happens when using Zap channels. Internal
calls work fine. No probs with the Snoms. No errors show on the * box
when the line
A good while back when installing 1.2 there were major issues with UK
callerid. Asterisk 1.2 didn't recognise the callerid correctly because
of the way BT sent the information. Sometimes before the first ring or
just after. After applying a third party patch we got it to work. We
were afraid
In article [EMAIL PROTECTED],
Benjamin Jacob [EMAIL PROTECTED] wrote:
Also, I don't think my SIP gateway uses Silence suppression, because the
same SIP gateway
connections work fine with another Asterisk server.
OK, I think you need to home in on the differences between the server(s)
that
To the best of my knowledge, multi-core processors are not hyperthreaded
- certainly my Core 2 Quad processor isn't. I would expect a Core 2 Duo
to be the same.
Steve Totaro wrote:
On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis [EMAIL PROTECTED] wrote:
Every CPU core shows up as a
Steve Totaro wrote:
That is interesting. I have an intel C2D and I can only see two
procs, not four, is that normal? Are you sure what you are saying is
I believe Intel removed HyperThreading after it moved over to dual cores.
Doug
--
Ben Franklin quote:
Those who would give up
On Fri, Apr 25, 2008 at 03:02:14PM +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Benjamin Jacob [EMAIL PROTECTED] wrote:
Also, I don't think my SIP gateway uses Silence suppression, because the
same SIP gateway
connections work fine with another Asterisk server.
OK,
Andres wrote:
We have tested both and they work fine. The Sangoma is much easier to
install as it does not depend on any other driver, you just run
'setup-sangoma' and follow the instructions. You don't have to fiddle
with the linux kernel or zaptel or chan_misdn. It just works. Plus
Benjamin Jacob [EMAIL PROTECTED] wrote:
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],
Benjamin Jacob wrote:
One on my clients' machine had Asterisk 1.4.4. installed. The complained of
choppy Playback
of gsm files.
So scouring the internet gave me the solution
On Thu, Apr 24, 2008 at 4:45 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote:
There are much better solutions than doing a RAM drive. While it may
be stable (not in my experience, I advise using different servers for
different
On Fri, Apr 25, 2008 at 12:22 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Apr 24, 2008 at 4:45 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Wed, Apr 23, 2008 at 02:14:27PM -0400, Steve Totaro wrote:
There are much better solutions than doing a RAM drive. While it may
be
On 4/25/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT)
From: Steve Edwards [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Forking in Dialplan
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
That is interesting. I have an intel C2D and I can only see two
procs, not four, is that normal? Are you sure what you are saying is
I believe Intel removed HyperThreading after it moved over
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
RTC is available (and used) as of kernel 2.6.15 . The thing that has
changed in 2.6.13 is that the default of HZ became 250 (but still
tunable). So unless you build your own kernel, without using RTC you
would not really
Steve Totaro wrote:
My dual proc, dual core AMD boxen show as four procs. I guess the AMD
architecture uses Hypertheading (or whatever the equivalent is for
AMD, I assume Intel owns the rights to the name Hyperthreading).
Not that I'm aware of.
But I did find this article from back in
Dear all, I'm using Asterisk 1.4.13 with voicemail feature. When anybody
receive a voice message, he/she receives a mail with the audio attachment.
After that I dial the voicemail number and I hear the envelope message
that is correct (America/Argentina/Buenos_Aires) which is GMT-3, but
when I
I still hope someone would enlighten us by his experience in doing call
recordings without recording to RAM Drive.
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On Fri, 2008-04-25 at 18:48 +, Arthur wrote:
I still hope someone would enlighten us by his experience in doing
call recordings without recording to RAM Drive.
I can't speak for Steve's solution (as I'm not sure exactly what he's
doing) but I could take a stab in the dark and guess that
i can only think of an asterisk box the right dialplan.
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he's capturing the audio at the network layer
i'd better stay with my 3Gigs RAM drive
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On 4/25/08, Jared Smith [EMAIL PROTECTED] wrote:
On Fri, 2008-04-25 at 18:48 +, Arthur wrote:
I still hope someone would enlighten us by his experience in doing
call recordings without recording to RAM Drive.
I can't speak for Steve's solution (as I'm not sure exactly what he's
When I initiate a call from the console (Console/dsp) to a local SIP
extension, asterisk uses up 100% of the CPU until the extension
answers. It happens when using .call files or the manager API. My
examples are for the manager API, but .call files perform the same way.
Here is the 100% CPU
On Friday 25 April 2008 15:23:05 Chris Elliott wrote:
If I reverse the situation it gets a little better. Asterisk doesn't
use 100% of the CPU, but until SIP/exten-20 answers, the manager
interface doesn't respond. So I can't hangup the line using the manager
API if SIP/exten-20 doesn't
Requirement: Monitor the QOS for the SIP phones connecting to the voip server.
Ideal solution: Browder based reporting software that I can install on
the asterisk server (I use freepbx) and when I connect to this
reporting engine it gives me the Jitter loss, packet loss and latency
for each of
Plus that originate is going to call the sip device, and upon answer
connect it to extension 0 in the internal context, is that what you wanted?
Tilghman Lesher wrote:
On Friday 25 April 2008 15:23:05 Chris Elliott wrote:
If I reverse the situation it gets a little better. Asterisk
On Fri, Apr 25, 2008 at 2:55 PM, Vikas [EMAIL PROTECTED] wrote:
B. Network between the SIP endpoints and VOIP server: The Indian
office has 5 different ISPs giving the internet connection. Each ISP
has a different packet loss latnecy and Jitter and these change over
time. So I want a way
Hi
I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.
zapata.conf
echocancel=64
rxgain=0
txgain=0
when i place a call o receive a call, I finish a
You might want to begin with tuning your rxgain and txgain settings... there
are a few methods for doing this on the internet, unfortunatly nobody can give
you exactly values to use for tx/rxgain as they will be not only specific to
your install, but specific to every single analog line you
On Fri, Apr 25, 2008 at 10:55 PM, Vikas [EMAIL PROTECTED] wrote:
Requirement: Monitor the QOS for the SIP phones connecting to the voip
server.
Ideal solution: Browder based reporting software that I can install on
the asterisk server (I use freepbx) and when I connect to this
reporting
Steve Totaro wrote:
On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
That is interesting. I have an intel C2D and I can only see two
procs, not four, is that normal? Are you sure what you are saying is
I believe Intel removed HyperThreading
As is just about always the case, posting twice to the list within three
hours is not only unlikely to get a faster response, I would in fact
imagine it would /reduce/ your chances of getting a response at all.
lotusscript wrote:
A good while back when installing 1.2 there were major issues
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