On Wed, Oct 03, 2007 at 10:48:37AM -0700, Steve Edwards wrote:
> I have an asterisk process that is consuming over 100mb (according to
> "top"). "Show channels" says "167 active channels and 53 active calls."
So you have 167 channels. There's a thread for each, with stack and all.
>
> It's an o
Hi Kevin,
Thanks for the answer - Hopefully this feature will be available some
day. My opinion is, look for a transcoder only if the two (or more)
parties does not offer any matching codec.
Good to hear it is being worked on
Best regards,
Ondrej
Kevin P. Fleming wrote:
> Ondrej Valousek wro
Nick Richardson a écrit :
> What if you don't use or want to use bristuff? We use Digium PRI cards
> and don't need any of the BRIstuff
As was said in previous posts, you don't need the full bristuff, just
res_watchdog.
Regards,
--
Jean-Denis Girard
SysNux Systèmes Linux
Le vendredi 05 octobre 2007 à 12:13 +1000, Nick Richardson a écrit :
> What if you don't use or want to use bristuff? We use Digium PRI cards
> and don't need any of the BRIstuff
yep i know ... but res_watchdog is part of bristuff. My tarball have
just the res_watchdog module with a makefile.
Hello,
This is a update on the current status of Asterisk in Debian.
Apologies for the really long mail, it is targetted both to users and
maintainers :)
I'm Ccing asterisk-users as a one-time thing; users that are interested
can subscribe to our list[1] for updates to prevent noise on a
non-Debia
Steve Totaro wrote:
> Steve Edwards wrote:
>> I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard
>> TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards.
>>
>> Each "group" of T1's have the primary D on 24 and the secondary D on 96.
>>
>> The first server (ts20) an
What if you don't use or want to use bristuff? We use Digium PRI cards
and don't need any of the BRIstuff
On 10/4/07, Sylvain Boily <[EMAIL PROTECTED]> wrote:
> Hi Nick,
>
> For using ISDNGuard, you can using res_watchdog from the bristuff patch.
> I attach for you a version who have patched
Hello,
> >> I am trying to use PHP to reload the extensions in an Asterisk
> >> installation. I keep getting this error:
> Easiest way without compromising security or changing permissions. Use
> the AMI.
>
> 1. Download phpagi (Just google it)
> 2. Use it to connect to the Manager interface
> 3
On Thu, Oct 04, 2007 at 04:10:09PM -0400, Arpit Mehta wrote:
> Hi all,
>
> I was looking at a way to add the caller id to the outgoing calls (which are
> made using .call files) using asterisk. Any ideas how to do this ?
> Currently I get 'Unknown' number displayed on my phone when asterisk makes
Also, so TFN's do not answer the line. Airline TFN are famous for doing
this.
Arick Davis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, October 04, 2007 2:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Su
The problem here, if it's a problem, is that it IS a POTS dialtone and
not asterisk's. So you can dial only what the telco lets you dial.
Al lists wrote:
> Here is how i overcome this problem,
> ignorpat => 9
> exten => 9*,1,Dial(ZAP/1/w)
>
> press 9* from your handset and after 1 second you h
Christoph Adomeit wrote:
> Hi,
>
> I have the following problem: I want asterisk to dial
> a chain of n-numbers until somebody picks up the line.
> I am using Digium E1 Hardware (zaptel) for dialing out.
>
> Dialing a Chain is basically no problem, I use somwthing like:
> dial(no1,50)
> dial(no2,
Mail Lists wrote:
> I am having some really strange problems calling from 2 asterisk boxes
> of mine. One is version 1.2.22 the other 1.2.18. The problem is
> identical on both boxes.
>
> When I try to call certain numbers (8006375410, for instance) the call
> rings and rings and rings. Eventually
bilal ghayyad wrote:
> Hi list;
>
> I need to run the command modprobe wctdm and whenever
> I write it, then it gives me the following message:
>
> FATAL: Module wctdm not found
> FATAL: Error running install command for wctdm
>
> So, do I have to run that command from specific path?
> Or what is
I am having some really strange problems calling from 2 asterisk boxes
of mine. One is version 1.2.22 the other 1.2.18. The problem is
identical on both boxes.
When I try to call certain numbers (8006375410, for instance) the call
rings and rings and rings. Eventually the receiving end will pick
Ondrej Valousek wrote:
> My problem is, that the phone offering g722 could do alaw as well.
> I expected asterisk should just chose alaw for the communication - no
> transcoding is necessary then...
That is not how Asterisk works, and is well known in the community as
something that users would l
At 11:10 10/4/2007, Alan Lord wrote:
>Hi,
>
>I am setting up an asterisk server for testing purposes and cannot get
>voicemail to work at all.
>
>My host OS is Linux From Scratch 6.3 and the asterisk software versions
>I built are zaptel-1.4.5.1 and asterisk-1.4.12.
>
>I am using the Ekiga
No, because then asterisk would be presented three arguments: '-rx',
'extensions', and 'reload' -- as 'extensions' is not a command by
itself, and the 'reload' appears superfluous to asterisk, this would not
work as desired.
Asterisk needs to be presented two arguments - the first is '-rx', the
Hi list;
I need to run the command modprobe wctdm and whenever
I write it, then it gives me the following message:
FATAL: Module wctdm not found
FATAL: Error running install command for wctdm
So, do I have to run that command from specific path?
Or what is the problem?
Any help?
Regards
Bilal
Arpit Mehta wrote:
> Thanks a lot guys. I got my answer from someone. :)
Not that I'm interested in this specific issue, but usually
it won't hurt to share your solution with the other list
members (and archives).
Doesn't apply to this case maybe.
Cheers,
Philipp Kempgen
--
amooma GmbH -
Alan Lord wrote:
> sip.conf
> =
>
> [100]
> type=friend
> callerid=Alan Lord
> secret=**
> qualify=yes ; Qualify peer is no more than 2000 ms away
> nat=no ; This phone is not natted
> host=dynamic ; This device registers with us
> canreinvite=no
Arpit Mehta wrote:
> I was looking at a way to add the caller id to the outgoing calls (which are
> made using .call files) using asterisk. Any ideas how to do this ?
Add
Callerid: Name <123>
to the call file.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://ww
On Thu, 2007-10-04 at 16:10 -0400, Arpit Mehta wrote:
> exten => _.,4,Set(CALLERID(all)=Joe <911>)
> exten =>
> _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/)
>
You need to set the Caller ID in the call file itself. The sample call
file (sample.call) in the Asterisk so
On 10/4/07, Arpit Mehta <[EMAIL PROTECTED]> wrote:
I was looking at a way to add the caller id to the outgoing calls (which are
> made using .call files) using asterisk. Any ideas how to do this ?
> Currently I get 'Unknown' number displayed on my phone when asterisk makes
> an outgoing call.
>
Hi,
I am setting up an asterisk server for testing purposes and cannot get
voicemail to work at all.
My host OS is Linux From Scratch 6.3 and the asterisk software versions
I built are zaptel-1.4.5.1 and asterisk-1.4.12.
I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk
Thanks guys. No need to reply. I got my answer from someone.
On 10/4/07, Arpit Mehta <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>
> I was looking at a way to add the caller id to the outgoing calls (which
> are made using .call files) using asterisk. Any ideas how to do this ?
> Currently I get 'Unkno
Thanks a lot guys. I got my answer from someone. :)
On 10/4/07, Arpit Mehta <[EMAIL PROTECTED]> wrote:
>
> Also what are the ways if any to set this DNIS or RDNIS information ?
>
> Regards
>
> Arpit
>
> On 10/4/07, Arpit Mehta < [EMAIL PROTECTED]> wrote:
> >
> > Hi Asterisk Users,
> >
> > I was
Hi all,
I was looking at a way to add the caller id to the outgoing calls (which are
made using .call files) using asterisk. Any ideas how to do this ?
Currently I get 'Unknown' number displayed on my phone when asterisk makes
an outgoing call.
Also using something like this is not working as it
Hello list
I am new in this list.
Before I wrote this email, i search with "google" and in the list
arichves for the question.
I look for a possibility to install FXO ports not over RJ11 Ports. I
will install the Ports by LSA+ Patch panel. Someone an idea ore link?
Thanks for help.
Bye MZ
PS: P
I never thought very useful to search in this
http://www.digium.com/en/docs/misc/compatibility_notes.php page, as it
looked rather static for holding such compatibility issues.
You proved me I was wrong.
Taking this e1000 driver issue as an example, this
http://sourceforge.net/projects/e1000 proje
Dear All,
my client wants a asterisk pbx with 30 FXO & 30 FXS analogue ports, please
suggest if sangoma A400 is a good option for that. Also please suggest
server hardware.
regards,
Umair
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On Wed, Oct 03, 2007 at 09:10:58PM -0500, Moises Silva wrote:
> If you are running the script from a web server, the script gets
> executed with the web server process permissions, hence, probably does
> not have access to /var/run/asterisk.ctl.
>
> You can give permissions to your web server, or
Well I know.
My problem is, that the phone offering g722 could do alaw as well.
I expected asterisk should just chose alaw for the communication - no
transcoding is necessary then...
Please help.
Thanks,
Ondrej
Kevin P. Fleming wrote:
> Ondrej Valousek wrote:
>
>
>> [Sep 20 10:14:32] WARNING[
Also what are the ways if any to set this DNIS or RDNIS information ?
Regards
Arpit
On 10/4/07, Arpit Mehta <[EMAIL PROTECTED]> wrote:
>
> Hi Asterisk Users,
>
> I was wondering why a call that is received from Asterisk shows a caller
> ID 'Unknown' . So here is the scenario,
>
> 'A' calls 'Ast
Just forget it about the 1.2 mantra, it's not going to happen. Focus your
energy elsewhere.
Lot's of bug fixes are good. Even Cisco comes out with regular bug fixes
for IOS. Open source just makes things more visible.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sen
There are other Gigabit SIP phones from Nortel and Avaya, if my memory
serves me right.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Hi,
I have the following problem: I want asterisk to dial
a chain of n-numbers until somebody picks up the line.
I am using Digium E1 Hardware (zaptel) for dialing out.
Dialing a Chain is basically no problem, I use somwthing like:
dial(no1,50)
dial(no2,50)
dial(no3,50)
However, If no1 is not re
Hi Asterisk Users,
I was wondering why a call that is received from Asterisk shows a caller ID
'Unknown' . So here is the scenario,
'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'.
'Asterisk' calls 'B'. 'B' gets joined to the same conference also.
'B' somehow receives the caller ID 'U
-- Forwarded message --
From: Tzafrir Cohen <[EMAIL PROTECTED]>
Date: Oct 4, 2007 12:56 PM
Subject: Re: [asterisk-dev] chan_h323 and chan_oh323 compatibilities
To: [EMAIL PROTECTED]
Hi
On Thu, Oct 04, 2007 at 11:46:30AM -0300, Caciano Machado wrote:
> I'm receiving a lot of warni
Hi Mujtaba,
We have a multi-tenant version of our Asterisk based management and end-user
software called Thirdlane PBX Manager. You can see a demo of a single-tenant
version on our web site http://www.thirdlane.com/pbxmanager.htm the
multi-tenant adds tenant and DID management, and allows to pa
On Oct 4, 2007, at 8:39 AM, Steve Totaro wrote:
Try searching using MGCP which is what Megaco evolved into.
http://www.voip-info.org/wiki-Asterisk+MGCP+channels
Thanks,
Steve Totaro
Too bad the MGCP channel isn't the full implementation.
/b
___
Floyd wrote:
> Hi all,
> I've been searching for a while and haven't found if
> asterisk supports already or if it's going to support
> h.248.
>
> thanks
> Eve
>
>
Try searching using MGCP which is what Megaco evolved into.
http://www.voip-info.org/wiki-Asterisk+MGCP+channels
Thanks,
Ste
Hello all, I need a little help to check the state of the line from
asterisk on aa TDM400P because when the telco lines goes down, asterisk get
that line for outgoing calls. There is a way to check it out?
And when all lines are busy to do outgoing calls how can i do to callback
the people th
On Thursday 04 October 2007 07:07:47 Barzilai Spinak wrote:
> All this discussion is pointless. As pointless as the discussion of
> assembly versus high-level languages decades ago.
As one of the main architects, I don't find this discussion pointless. My
personal opinion of AEL is that it's comi
Hi all,
I've been searching for a while and haven't found if
asterisk supports already or if it's going to support
h.248.
thanks
Eve
¡Sé un mejor ambientalista!
Encuentra consejos para cuidar el luga
Hi Mark,
Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark Michelson:
> GNUbie wrote:
> > Hello all,
> >
> > Is it possible to store, read and write configuration files in an
> > SQLite3 database instead of using the configuration files inside the
> > /etc/asterisk/ directory? If it is then
All this discussion is pointless. As pointless as the discussion of
assembly versus high-level languages decades ago.
Except most people rooting for "extension.conf" don't even have the
technical and conceptual amplitude to understand what they are talking
about... they just want some telephony
Christian Weeks wrote:
> Hi
> I've had an asterisk setup for the past 15 months, based on the debian
> asterisk packaging. Until late August of this year, I had no problems
> once initial setup was complete- the system worked essentially
> flawlessly.
>
> Since August I have been having exceedingly
Dear Walt;
Maybe I did not understand any thing from below :) -
Are you talking about configuration to be done on the
Telephone device is self or on the AVAYA server it
self? If it is on the telephone device, so how you
will give a second dial tone and you do not know if
there is available channe
Steve Edwards wrote:
> On Wed, 3 Oct 2007, Steve Totaro wrote:
>
>
>> Kevin P. Fleming wrote:
>>
>>> Steve Edwards wrote:
>>>
>>>
>>>
[trunkgroups]
trunkgroup = 1,24,96
spanmap = 1,1,0
spanmap
Dear Mojo;
That is primary fine, but there are two issues looking
for help about them:
1) Based on your below example (dialing *4*18005551212
to select channel 4), the question is how to give
second dial tone just after dialing the *4*
(indicating the channel was captured)?
2) How to let this se
Hi all,
i just wanted to know if any one has done any multi-tenant version of the
asterisk.
thanks
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On 10/4/07, chawki hammoud <[EMAIL PROTECTED]> wrote:
> I have a digium tdm04B. One fxo module always gives a
> faulty busy signal when a call comes in. I swaped
> places and re-compiled zaptel with no change. Is this
> a faulty fxo or configuration issue.
Hi,
I suggest you post your related zap
This Friday at 12:30 PM EDT
We hope to hear more about Astricon and the 1.6 version. A UK legal
professional, John Halton of Cripps Harries Hall LLP, joins us to
discuss how the law is coming to terms with VOIP. We also expect a
visit from Arick of IPKall about what's cooking with them. Most of
al
Hi Chawki,
it is not uncommon that FXO or FXS modules do not work even if no error
message is shown when wctdm module is loaded.
Have you tried to replace/swap your FXO modules?
Giorgio
chawki hammoud wrote:
> Hi:
>
>
> I have a digium tdm04B. One fxo module always gives a
> faulty busy signal w
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