On Tue, Mar 04, 2008 at 03:05:43AM +, Ex Vito wrote:
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system. How
can I tell if the card has a TigerJet chipset ?
Off-topic:
Are those just the cards that show
Quoth: OCG Technical Support [EMAIL PROTECTED]
Although we've programmed the softkeys per the manuals, they seem to have no
effect (just dead). For example, our 57i is setup like this:
I had similar problems and ended up using the speeddial inband
functionality. FWIW, my 57i's setup like so:
So far these people let me know there are going to be there, who else is
going and wants to do some networking
Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com -
wednesday / thursday.
Tan Aksoy - Telappliant - wednesday / thursday
Antoine Megalla - SAND - wednesday /
I've tried this branch, but no luck again Dialing out work just fine,
but no incoming calls are recognized. With absolutely no message in logs
:-( I also have seen this mentioned here:
http://bugs.digium.com/view.php?id=12099
Martin
- Original Message -
From: sean darcy [EMAIL
I may be asking the wrong question, but if you want to capture the
input and pass it to another process why not use the read application
and store the input in a variable? Could you not pass that variable
and use the curl function or an AGI to post it?
On Mon, Mar 3, 2008 at 11:05 PM, Prashant
Having recently worked with the latest version here are my thoughts.
1-3 The only out of the box method to configure and manage switchvox
is the GUI. They are using realtime but manually configuring the
system is not available by default. SSH access is not available by
default and the root
You should stop asterisk first, otherwise modules are still in use
And then
/etc/init.d/zaptel stop
will remove modules.
Martin
- Original Message -
From: Paul Hales [EMAIL PROTECTED]
Sent: 28. února 2008 19:35
Subject: Re: [asterisk-users] Problems with removing zaptel
Hi Ian,
I will try this workaround, I'll be trying to get this to work with the
chan_local solution, if I have success I'll let you know, thanks...
On Wed, Mar 5, 2008 at 2:37 AM, Ian [EMAIL PROTECTED] wrote:
Hi Raul
I have bypassed my Grandstream's transfer function, by enabling *2
hi,
can you recommend cleansimplestable solution for incoming call popup
(in browser)?
i'm using flash operator panel now
but i want something without flash (maybe something in AJAX?)
thanks
---
Marek Cervenka
===
Does any one know how to change astmanproxy to be able to
a) compile without optimisations
b) dump a core
I've had it crash several times over the past couple of months, but
there is no way to debug what's going on.
I like the way a core is produced when (if!) * crashes, and would like
to
On Tue, Mar 04, 2008 at 01:53:07PM +, Julian Lyndon-Smith wrote:
Does any one know how to change astmanproxy to be able to
a) compile without optimisations
b) dump a core
I've had it crash several times over the past couple of months, but
there is no way to debug what's going on.
Has anyone here done any work on clustering Meetme conferences over
multiple Asterisk boxes? The scenario I am thinking of is where there are
two or more boxes connected to a set of PRIs that all answer to the same
PSTN number, and where it's not possible to know in advance on which box
a call
*bump*
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 21, 2008 11:55 AM
Will the Page() command handle 200 SIP devices?
How much time does it take for ALL 200 devices to be ready
to receive audio?
Jerry
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
Hello,
We have done this using IAX trunks between Asterisk servers to connect
a PRI line on server A with a meetme room on server B. We have had
hundreds of participants in meetme rooms across a dozen Asterisk
servers using this method.
Not knowing your setup I'm not sure if this would work
Martin wrote:
I've tried this branch, but no luck again Dialing out work just fine,
but no incoming calls are recognized. With absolutely no message in logs
:-( I also have seen this mentioned here:
http://bugs.digium.com/view.php?id=12099
I think it might work for you if you try the
On Tuesday 04 March 2008 09:45:38 Mike Hammett wrote:
*bump*
If people don't know, they don't know. There is no need to repost your query
10 days later. Not that many more people have signed up, and those who have
signed up are unlikely to be able to answer your question.
The only thing that
Hi Matt, thanks for your reply.
In article [EMAIL PROTECTED],
Matt Florell [EMAIL PROTECTED] wrote:
Hello,
We have done this using IAX trunks between Asterisk servers to connect
a PRI line on server A with a meetme room on server B. We have had
hundreds of participants in meetme rooms
I previously posted about this problem and received suggestions
involving turning off echo cancellation. As far as I can tell, echo
cancellation is already disabled on this channel, so I'm back.
What I've got is a small home setup with a single four-port Digium card:
Module 0: Installed -- AUTO
Hello,
I have actually done this both ways, with many small conferences and
few large conferences.
The best example of both is the voice_lab feature that is included
with VICIDIAL(although not very well documented). What this feature
does is it has students log into individual meetme rooms and
Greg Woods wrote:
The problem is that PPP dialout does not work; pppd eventually fails
with NO CARRIER. Dialout works fine if I connect the modem straight
through to the wall plate, the problem only happens when the modem is
connected through Asterisk and the Digium card. The same modem and
I couldn't figure it out on my own. I tried to purchase a Smartnet for the
phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have someone
remote into my network and upgrade my SCCP 7960 to the latest SIP firmware?
--
Mike
I was doing it because of the volume on the server. It is very easy to miss
a message or 10 or 100 on a list of this traffic.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users
Mike,
I might be able to help you. Contact me offline. s-h-a-n-e-b-AT-m-e-
t-r-o-s-t-a-t-.-.n-e-t
Shane
On Mar 4, 2008, at 1:12 PM, Mike Hammett wrote:
I was doing it because of the volume on the server. It is very easy
to miss
a message or 10 or 100 on a list of this
I've been able to run low speed modems through a SIP ATA and an IAX
trunkbridged by asterisk.
I would assume that in this day and age any modem application through
asterisk is probably for either a remote console or some sort of control
system. Either way using SIP and a very low speed
Mike Hammett wrote:
I couldn't figure it out on my own. I tried to purchase a Smartnet
for the phone, but the original 7960 is not supported.
Is it technically possible and if so, what would it cost me to have
someone remote into my network and upgrade my SCCP 7960 to the latest
SIP
Hi, all
My Asterisk uses a Digium TE120Pand I would like to add an Astribank
zaptel_hardware sees is, but I cannot get it working
pbx:~# zaptel_hardware
Argument IRQ isn't numeric in numeric comparison (=) at
/usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114.
usb:005/002 xpp_usb-
I just connected an Asterisk server to an Avaya Pbx using the
instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk
+Avaya
Everything seems to be working as I can send and receive calls. The
only detail I am having a problem with is that when an extension on the
That I am. I'll contact you off list.
--
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Sigma Networks [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday,
On Tue, 2008-03-04 at 12:05 -0600, Kevin P. Fleming wrote:
Greg Woods wrote:
The problem is that PPP dialout does not work
From:
Kevin P. Fleming
[EMAIL PROTECTED]
it is not likely that you will be able to accomplish what you
want using an analog interface
On Tue, Mar 4, 2008 at 7:54 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Mar 04, 2008 at 03:05:43AM +, Ex Vito wrote:
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system. How
can I tell if the
I just upgraded zaptel to 1.4.9.2 (and rebuild everything, of course)
but no improvements.
zaptel_hardware's output is the same and genzaptelconf
I forgot to mention that zapconf fails too...
pbx:~# zapconf
Argument IRQ isn't numeric in numeric comparison (=) at
Carlos Chavez wrote:
Avaya calls an Asterisk extension, I only get the name part of the CID,
not the number. From Asterisk to the Avaya we can see both the name and
the number displayed on the screen.
We had the same issue with one of our older nodes. The only way we were
able to get
Carlos Chavez wrote:
Avaya calls an Asterisk extension, I only get the name part of the CID,
not the number. From Asterisk to the Avaya we can see both the name and
the number displayed on the screen.
[bc-indianapolis]
exten = _41XX,1,Gosubif($[${CALLERID(number)} = ]?get-cid,s,1:2)
Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system
to work with *?
I'm not getting inbound or outbound calls to work. (Inbound caller gets
line busy tone.)
SETTINGS FROM MITEL:
I built a Crossover cable and connected it like this:
PSTN--T1--ASTRISK--T1--OLD MITEL
On Tue, Mar 04, 2008 at 07:04:15PM +, Andres Jimenez wrote:
Hi, all
My Asterisk uses a Digium TE120Pand I would like to add an Astribank
zaptel_hardware sees is, but I cannot get it working
What version of Zaptel is this?
pbx:~# zaptel_hardware
Argument IRQ isn't numeric in
Is there a way to crank the volume on Polycom speaker phones? The 430
and 4000 that I have are quieter than expected. The volume on the
device is turned all the way up, but are there firmware options that
can be set?
___
-- Bandwidth and Colocation
I would strongly recommend ESF/B8ZS. If you have a RED alarm that means
the device does not see a line connected to it -- check cabling.
Mark Best wrote:
Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system
to work with *?
I'm not getting inbound or outbound calls to
I am trying to get a console/dsp application going with
1.4.18 and not hearing any audio. In the CLI I see the call coming in,
I see the Dial(Console/dsp)
I see auto answered
I see ALSA default
but I hear no audio.
What can I do to tell what is happening here.
I have in modules.conf:
noload
Eric,
Looks like my Mitel does support ESF; I'll try changing it. One question
before I do. If I change it right now, do you suppose my Telco
automatically adapt?
TELCO---T1---MITEL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent:
On Tue, 2008-03-04 at 12:34 -0800, Mark Best wrote:
Does anyone have any experience getting a Mitel SX-200 EL/ML PBX
system to work with *?
No, but I'll make a few comments anyway. I'm not sure if they'll help
you at all, but maybe it'll help explain a few things to someone else on
the list.
Yes, they are listed in the Admin Manual for the Polycoms
Peter Hessler wrote:
Is there a way to crank the volume on Polycom speaker phones? The 430
and 4000 that I have are quieter than expected. The volume on the
device is turned all the way up, but are there firmware options that
can
On Tue, 2008-03-04 at 13:07 -0800, Mark Best wrote:
Looks like my Mitel does support ESF; I'll try changing it. One question
before I do. If I change it right now, do you suppose my Telco
automatically adapt?
I've never seen a telco automatically adapt to anything, especially
signaling changes
Jared,
Thanks for the help.
There are other T1 trunks (using AMI,D4) going to different departments;
however the trunk line talking to the Telco is what it is.
I do have the ability to change it to ESF, so I will try changing it -
do you suppose my Telco will automaticly adapt if I do? (I'm
Hi all,
I've been working on debugging a bit of a custom dialplan system, and
seem to have run into some issues on our development server. Hopefully
someone can give me some pointers on this one! =)
In a nutshell, we have a hangup extension that's being triggered to feed
data back to an api
The telco will never change the line coding or framing without a change
request being sent to your account rep.
Mark Best wrote:
Eric,
Looks like my Mitel does support ESF; I'll try changing it. One question
before I do. If I change it right now, do you suppose my Telco
automatically adapt?
Steve,
Thanks for your input but please allow me to clarify that I'm not a
noob. I'm perfectly capable of looking up the definition of a protocol
error just the same as you... And I had already done that before I
posted the question. I think you misunderstood what I posted. The
switchtype in the
Jerry Geis wrote:
I am trying to get a console/dsp application going with
1.4.18 and not hearing any audio. In the CLI I see the call coming in,
I see the Dial(Console/dsp)
I see auto answered
I see ALSA default
but I hear no audio.
What can I do to tell what is happening here.
I have
sean darcy wrote:
ekiga registration fails. I've set nat = yes ( also blank ) and i've set
externip. Anybody have a sip.conf that works?
Here's the sip debug:
Reliably Transmitting (NAT) to 86.64.162.35:5060:
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP
Hi! I've managed to make Asterisk 1.2.26 work with CCM 4.2 using SIP.
With H.323 I had some issues.
I'm working to integrate Asterisk with Unity (CCM Voicemail) with
VPIMv2. To make it I am developing something that make Asterisk VPIM
capable.
Hope it helps.
Greets.
On Thu, Feb 28, 2008 at
I just managed to put in a TE410 card in an Asterisk box to work with
OnRamp 20(E1 downunder). I am able to dial in but was not able to dial
out.
Can anyone offer me some advice please?
In my extensions.conf, I just put in:
[default]
...
exten = 0,1,Dial(Zap/g1)
and I get this on the console
Thanks Bruce.
exten = 0,1,Dial(Zap/g1/0) does not work.
exten = 0,1,Dial(Zap/g1/${EXTEN:1)) works.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, 5 March 2008 4:54 PM
To: Asterisk Users Mailing List -
Hi Raul
Thank you, that would be apreciated, btw I have been talking to the
users, and they say the *2 transfers are easier than the built in
grandstream transfers. I also forgot to state that I had to set the
Grandstream to send DTMF via sip info, or else it will only work some of
the time.
John,
Try changing the entry in extensions.conf to Dial(Zap/g1/0). you need
to specify what the dial command should send on the ZAP channel.
On Tue, Mar 4, 2008 at 10:51 PM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I just managed to put in a TE410 card in an Asterisk box to work with
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