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
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
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 old
On Thu, Oct 04, 2007 at 02:15:12PM -0700, 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
What is the output of:
modinfo wctdm
Maybe you got the kernel version wrong
Doug wrote:
snip /
I have a single x100p card installed which seems to work - to a
fashion... Incoming calls are answered and the greeting is heard, but
the line hangs up instantly the message finishes. (A different problem
which I will investigate seperately unless someone has a quick
Hi Marco,
could you try again with different quality settings on your hardware fax?
I see 1728 x 540 in your spandsp log.
I think spandsp is only compatible with :
standard: width=1728 x length=1128
fine: width=1728 x length=2252
super-fine: width=1728 x length=4491
which are the 3 resolution you
Hello,
the bench that gives me a problem is the following:
asterisk1 + spandsp ==PSTN== asterisk2 + spandsp
1) i put a .call file inside Asterisk spool directory
2) the fax is sent successfully, I got this message in the sender's log:
[Oct 4 17:57:02] DEBUG[17610] app_txfax.c:
Ciao Tzafrir,
The only real benefit is if you can limit the permissions you give to
that specific manager user. But there's a limit to ohw useful this can
be. Even write=command alone allows changing the dialplan ('dialplan
add' / 'dialplan remove') and running an arbitrary command as the
In article [EMAIL PROTECTED],
Tilghman Lesher [EMAIL PROTECTED] wrote:
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
In article [EMAIL PROTECTED],
C F [EMAIL PROTECTED] wrote:
If you want this to work nicely dont settle for anything else than a
channel bank
So why have Digium bothered to market a TDM2400P then? :-S
Cheers
Tony
On 10/3/07, Thomas Kenyon [EMAIL PROTECTED] wrote:
Mojo with Horan Company,
In article [EMAIL PROTECTED],
Steve Totaro [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
I have a client who wants a Meetme box with 12 FXO ports, to connect
to Analogue lines coming from an Ericsson PBX.
It looks like I could do this with four different hardware configurations:
a)
Hello,
I'm new to LDAP.
I've read Device class exists (oid 2.5.6.14) in rfc2256.
I've heard a Pluggable Device sub-class (a device with a MAC address) also
exists though I can't find its OID at the moment.
1. Does any standard class specifically defines IP Phones or SIP hardphones
or ATAs or
Original Message
From: [EMAIL PROTECTED] (Tony Mountifield)
Date: Fri, October 05, 2007 4:05 am
To: asterisk-users@lists.digium.com
In article [EMAIL PROTECTED],
C F [EMAIL PROTECTED] wrote:
If you want this to work nicely dont settle for anything else than a
channel
Greetings list,
Over the last couple of years quite a few other open source PBX/softswitch
applications - Freeswitch, YATE, SER to name but a few - have sprung up. People
on this list have mentioned in passing different scenarios where they might be
a good asterisk alternative or addition.
Nothing from me is posting to the list either.
Julian
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On Fri, 2007-10-05 at 09:17 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Tilghman Lesher [EMAIL PROTECTED] wrote:
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
Doug Lytle wrote:
Nothing from me is posting to the list.
Sure, that comes though. I've been trying to post the below message:
Does anybody have some scripts that will pound an Asterisk system?
I wanted to see how many channels I could produce on my test system
before it would be die.
Hi Eric,
thanks for your hint.
Unfortunately it doesn't work, I just tested it. It seems
that at least in german T-D1 Mobile Network a mobile call
is answered even if that mobile is switched off.
On Thu, Oct 04, 2007 at 05:30:11PM -0500, Eric ManxPower Wieling wrote:
Christoph Adomeit wrote:
Well, I would put in logic so the person has to confirm by hitting 1
or something and otherwise no matter what the phone company does, you
will know. We have the same problem here and this is the only thing I
can think of to fix it.
on Friday 10/05/2007 Christoph Adomeit([EMAIL PROTECTED]) wrote
Tony Mountifield wrote:
Has anyone here had any experience of using the Intel S5000PAL server
board (supplied in Intel SR1500, SR1550 or SR2500 chassis) with Digium
cards?
We have a number of them and we have tested our PCI-X and PCI-E cards
with them without experiencing any problems. These
Julian Lyndon-Smith wrote:
Nothing from me is posting to the list either.
heh. Thought that this trick would work: it did for Doug.
I've been trying to send the email below for 3 days now !
I know this is probably going to ignite the flames again ..
I have looked at the recent threads
Has anyone here had any experience of using the Intel S5000PAL server
board (supplied in Intel SR1500, SR1550 or SR2500 chassis) with Digium
cards?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
Alan Lord wrote:
snip /
As I said originally, I can call the local echo extension set up on x611
and that works fine. When I do 'set rdp debug' I can see some messages
relating to RFC2833 packets being transmitted and asterisk seemingly
acknowledging their receipt.
I have tried
Hi all, I have a network with some asterisk in trunk with IAX2 and some
SIP/ZAP phone connect to this *.
In every call I need to use only alaw codec so in all conf file I have
set disallow=all and allow=alaw.
I try also to make some tuning of my environment removing unused codec
and application.
I've got some scripts designed to make about 700 calls/minute out of it
using the Manager API, if you like.
On Fri, 5 Oct 2007, Doug Lytle wrote:
Doug Lytle wrote:
Nothing from me is posting to the list.
Sure, that comes though. I've been trying to post the below message:
Does
Nothing from me is posting to the list.
*sigh*
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
--Bandwidth and Colocation Provided by
I would like to point out that G.722 is a really awesome codec for
wideband. Asterisk has some changes that will need to be made to
support variable audio rates. We did this in FreeSWITCH from the
start. I think Asterisk will be doing similar things to bridge an 8k
to 16k channel via
Brian West wrote:
open market and an open platform. Rhino makes hardware that plugs into
zaptel but yet I don't see their drivers in the zaptel repo... I don't
see many of the third party hardware drivers in the zaptel repo.
Those drivers would be there (as are the Xorcom XPP drivers) if
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when using g722.
G722 is free and in the clear. G722.1 and G722.2 are not.
We have the G722 code in FreeSWITCH donated to us by Steve
Underwood. What a great guy.
/b
On Oct 5, 2007, at
Kevin,
Thats good to know. I'll keep that in mind.
Thanks,
Brian
PS: did you ever talk to mark about zaptel.h ?
On Oct 5, 2007, at 8:12 AM, Kevin P. Fleming wrote:
Those drivers would be there (as are the Xorcom XPP drivers) if they
were properly submitted and met our coding
Tilghman Lesher wrote:
On Wednesday 03 October 2007 06:09:01 Peter Fern wrote:
Of course, I could be missing something obvious, please correct me if
that's the case.
I invite you to try it. You could make a lot of really smart people look like
fools if you're able to mix
Alex Balashov wrote:
I've got some scripts designed to make about 700 calls/minute out of it
using the Manager API, if you like.
Yes, please!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor
Bad translation - I understood awesome = bad :-)
I agree - That is why I wanted to start using it asap - hopefully the
full support in Asterisk will be here soon.
O.
Brian West wrote:
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when
But its way too heavy on the CPU.
/b
On Oct 5, 2007, at 8:34 AM, Tzafrir Cohen wrote:
But speex *Is* free. Including wideband.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or
Tzafrir Cohen wrote:
On Fri, Oct 05, 2007 at 08:12:34AM -0500, Brian West wrote:
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when using g722.
G722 is free and in the clear. G722.1 and G722.2 are not.
But speex *Is* free.
Sangoma has contributed to Asterisk in the past and they still do.
They also have contributed to Yate, FreeSWITCH and various other
software that is capable of using their hardware. This argument of
Digium vs Sangoma is very emotional for some. I see it as
competition is good and drives
Hi Brian,
No I do not need g722 that seriously - I just thought - we all in the
company have phones that support it, we are all on switched LAN (so
bandwidth is not a problem) - so why not use it?
At this point I would like to know why you think it is awesome? I know
the are some
On Fri, 2007-10-05 at 13:06 +0100, Julian Lyndon-Smith wrote:
Julian Lyndon-Smith wrote:
Nothing from me is posting to the list either.
heh. Thought that this trick would work: it did for Doug.
I've been trying to send the email below for 3 days now !
I know this is probably going
Dear Cohen;
Your help was great, now it is loading.
But as a favourite, could u please explain for me what
was you mean by the following:
FATAL: Error running install command for wctdm
This is caused by the silly post-install command for
wctdm in
/etc/modprobe.d/zaptel or /etc/modprobe.conf
On Fri, Oct 05, 2007 at 08:12:34AM -0500, Brian West wrote:
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when using g722.
G722 is free and in the clear. G722.1 and G722.2 are not.
But speex *Is* free. Including wideband.
--
Steve Murphy wrote:
Oh, Julian, I'd imagine what I'm about to say will fuel some flames!
Here's a fairly powerful argument for all you asterisk users, as to why
you
should purchase a Digium product vs. a Sangoma: Because Digium uses a
chunk
of the purchase money to support Asterisk. And
In article [EMAIL PROTECTED],
Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Has anyone here had any experience of using the Intel S5000PAL server
board (supplied in Intel SR1500, SR1550 or SR2500 chassis) with Digium
cards?
We have a number of them and we have tested
Send these questions to Asterisk-Users mailing list.
h323.conf
##
;
; Configuration file of OpenH323 channel driver
;
[general]
listenAddress=W.X.Y.Z ; local ip
listenPort=1720
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
Tzafrir Cohen wrote:
On Fri, Oct 05, 2007 at 07:41:38AM -0400, Doug Lytle wrote:
Use a second Asterisk system on a stronger machine (or several Asterisk
systems on several machines). You know how to program Asterisk, right?
I have some experience in that area, yes. At this moment I
On Oct 5, 2007, at 9:31 AM, Tzafrir Cohen wrote:
How many hardware vendors support g722.1 ? g722.2 ? How pleasent are
they to the CPU? How much does it cost them?
I think polycom does and both are very heavy on CPU.
Naturally I don't suggest to use speex/wb where there is enough
bandwidth
On Fri, Oct 05, 2007 at 07:41:38AM -0400, Doug Lytle wrote:
Doug Lytle wrote:
Nothing from me is posting to the list.
Sure, that comes though. I've been trying to post the below message:
Does anybody have some scripts that will pound an Asterisk system?
Use a second Asterisk
On Fri, Oct 05, 2007 at 09:45:06AM -0400, Julio Arruda wrote:
Tzafrir Cohen wrote:
On Fri, Oct 05, 2007 at 08:12:34AM -0500, Brian West wrote:
You can hear and understand someone much better with g722... more
emotion is transfered over the phone when using g722.
G722 is free and in
Hi,
I'm building a test asterisk server and building the latest kernel I got
to wonder if there are any specific recommendations about schedulers and
so forth for optimum performance.
There are a few areas that raise questions in my mind and I wonder if
anyone has any opinions/comments on
Brian West wrote:
Sangoma has contributed to Asterisk in the past and they still do. They
also have contributed to Yate, FreeSWITCH and various other software
that is capable of using their hardware. This argument of Digium vs
Sangoma is very emotional for some. I see it as competition
Ok.. will be there...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, October 04, 2007 12:50 PM
To: asterisk-users@lists.digium.com
Subject: asterisk-users Digest, Vol 39, Issue 12
Send asterisk-users mailing list
This is really a silly debate.
I used to buy Digium products until they let me down with all kinds of
quirky behavior with regards to echo, clicks, incompatible
motherboards and IRQ issues.
I read all the success and praises of Sangoma on this list and thought I
would give them a try. Guess
FYI, Sorry for crossposting to the users and commercial list but this is
really a HUGE development and deserves a large Asterisk audience.
Digium = Shrewd business moves as I predicted when they purchased Sokol,
it's like a game of chess!
This is your 1-day reminder for the event:
Event:
Brian West wrote:
I would like to point out that G.722 is a really awesome codec for
wideband. Asterisk has some changes that will need to be made to
support variable audio rates. We did this in FreeSWITCH from the
start. I think Asterisk will be doing similar things to bridge an 8k
to
Brian West wrote:
Sangoma has contributed to Asterisk in the past and they still do.
Which contributions are you talking about, exactly? I know that they paid
someone to write app_dictate a couple of years ago, but that is the only thing I
can think of that has come through since I have been
I think Lee Howard nailed it.
/b
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Matthew Fredrickson wrote:
Not to ignite any fires, but I don't think I've *ever* knowingly
received a patch to libpri or chan_zap from them. And I've fixed a few
protocol related bugs in libpri for people with Sangoma cards. It'd be
nice if they at the very least supported the protocol
All,
I'm having an issue deploying softphones into my DUNDi/regcontext
setup. My current design is that all SIP users get registered into a
sipregistration context in the sip.conf. I then have a dialplan
function that includes that and does the dial:
include = sipregistration
exten =
Thomas Kenyon wrote:
Steve Murphy wrote:
Oh, Julian, I'd imagine what I'm about to say will fuel some flames!
Here's a fairly powerful argument for all you asterisk users, as to why
you
should purchase a Digium product vs. a Sangoma: Because Digium uses a
chunk
of the purchase money to
Lee Howard wrote:
Any Digium competitor is immediately on unequal footing with respect to
Asterisk due to the dual-license and requisite disclaiming of
contributions. You're asking those competitors to contribute not only
to the open-source Asterisk, but also to contribute to Digium's ABE
On Fri, 2007-10-05 at 08:05 -0500, Brian West wrote:
Sangoma has contributed to Asterisk in the past and they still do.
They also have contributed to Yate, FreeSWITCH and various other
software that is capable of using their hardware. This argument of
Digium vs Sangoma is very emotional for
I am curious if anyone has used the Asterisk Appliance in an
install. Where you happy with it?
I have done a couple installs with 250-300 phones, but I have another
one coming up that is only 7 phones. I thought I would give the
Asterisk Appliance a try.
Is it able to echo cancel on
Alan Lord wrote:
I'm building a test asterisk server and building the latest kernel I got
to wonder if there are any specific recommendations about schedulers and
so forth for optimum performance.
There are a few areas that raise questions in my mind and I wonder if
anyone has any
The settings I use are as follows (with explanations):
SLAB Allocator (SLAB or SLUB?)
SLAB (the default).
I don't know enough about the differences between them to say whether one is
better than the other.
Tickless System (?)
Enabled.
Seems to be the default setting. Disabling it doesn't
OK, I found the answer to my echo question (32ms).
But, has anyone used it? Feelings?
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE
Lee Howard wrote:
The report appears to have been reaped from Mantis, but I was involved
with a contribution from OpenVOX for zaptel, and from my perspective it
looked like the Digium staff involved killed it and never gave any
indication that the contribution would be accepted.
I assume
On Fri, 2007-10-05 at 12:23 -0400, Forrest Beck wrote:
Is it able to echo cancel on the FXS/FXO cards?
Yes, it has has built-in echo cancellation in hardware.
--
Jared Smith
Community Relations Manager
Digium, Inc.
___
--Bandwidth and Colocation
On Fri, 2007-10-05 at 12:26 -0400, Matthew J. Roth wrote:
Alan Lord wrote:
I'm building a test asterisk server and building the latest kernel I got
to wonder if there are any specific recommendations about schedulers and
so forth for optimum performance.
There are a few areas that
I just got the 2nd edition Asterisk book from O'Reilly, and was
surprised
to find nothing in there about AEL, except a mention of extensions.ael
on
page 471.
This is too bad. A preliminary chapter, an intro into AEL, why it's
valuable, etc. would have been very welcome. Even an appendix of
On Fri, 2007-10-05 at 11:32 -0400, Steve Totaro wrote:
I used to buy Digium products until they let me down with all kinds of
quirky behavior with regards to echo, clicks, incompatible
motherboards and IRQ issues.
(Let me take off my Digium hat for a minute and speak as a community
member,
On Fri, Oct 05, 2007 at 12:26:37PM -0400, Matthew J. Roth wrote:
(
1. The kernel version must be at least 2.6.13
3. The kernel must be configured to provide RTC interrupts
4. The kernel must be configured with enhanced real time clock support
)
Or:
2. The kernel must be configured
I wonder:
On Fri, Oct 05, 2007 at 05:33:07PM +0100, Chris Bagnall wrote:
The settings I use are as follows (with explanations):
Tickless System (?)
Enabled.
Seems to be the default setting. Disabling it doesn't seem to have any
noticeable effect on system performance.
Preemption
I disagree with any argument for or against Digium in support of Asterisk as
much as I do for or against Sangoma or Rhino or one of the Chinese knock
offs in support of Asterisk. Digium uses the open source community to
create better commercial software products and their licensing policies
Hi, all:
I think everybody is entitled to their biases, and I have to say that --
far from seeing this as a flame-war or otherwise tedious -- I think it's
great that we're having this discussion and getting open and honest
input from Digium staffers. We want to hear your thoughts and feelings
Like fixing the poor design of the TDM400P and TE110 with the newer cards
that advertise VoiceBus. For a company that supposedly embraces the open
source philosophy I don't think Digium has been very forthcoming with what
they are doing so they should not be surprised by any apparent lack of
Michael Collins wrote:
I just got the 2nd edition Asterisk book from O'Reilly, and was
surprised
to find nothing in there about AEL, except a mention of extensions.ael
on
page 471.
This is too bad. A preliminary chapter, an intro into AEL, why it's
valuable, etc. would have been very
I was completely against the dual licensing in the beginning. But now, I'm
leaning more towards understanding it and the importance of it, especially
as it related to US Patent laws. We're going to find that everything is
patented in the US. This is going to be the demise of open source in the
Jared Smith wrote:
On Fri, 2007-10-05 at 11:32 -0400, Steve Totaro wrote:
I used to buy Digium products until they let me down with all kinds of
quirky behavior with regards to echo, clicks, incompatible
motherboards and IRQ issues.
(Let me take off my Digium hat for a minute and speak as
On Fri, 2007-10-05 at 09:17 +, Tony Mountifield wrote:
I just got the 2nd edition Asterisk book from O'Reilly, and was surprised
to find nothing in there about AEL, except a mention of extensions.ael on
page 471.
That's because we were rushed on the book, and none of the authors has
On Fri, 2007-10-05 at 08:40 -0700, Michael Collins wrote:
This is too bad. A preliminary chapter, an intro into AEL, why it's
valuable, etc. would have been very welcome. Even an appendix of a few
pages with examples and references to on-line documentation would have
been helpful. I don't
I've been considering replacing a PRI with SIP or IAX trunks. The
monthly cost difference is marginal, but it would save a bit on the
hardware side and soft trunks would be easier to manage. I can't help
but wonder what I would be giving up? I'd like to hear some lessons
learned from those
Without knowing more, Why fix what isn't broken?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield
Sent: Friday, October 05, 2007 1:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Replace full PRI
callerid can be set using the variables provided in asterisk,
$CALLERID both in .call file (if you are using one) and the
extensions.conf file.
Regards
Arpit
On 10/4/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Arpit Mehta wrote:
Thanks a lot guys. I got my answer from someone. :)
Not
The idea of adding every network device to directories was popular five years
ago, but didn't really move beyond iPlanet, NDS, and AD marketing and a few
enterprise network management solutions.
Unless you have a specific application this sounds like more trouble than it's
worth.
Hi friends.
How do I can forwarding a call to any extension in system A (running Asterisk)
if it did not answear to the voice mail in the system B (other Asterisk)? so
later users of system A can check the remote messages.
Do i have to create each extension on B? Which are the configuration of
Jim Canfield wrote:
I've been considering replacing a PRI with SIP or IAX trunks. The
monthly cost difference is marginal, but it would save a bit on the
hardware side and soft trunks would be easier to manage. I can't help
but wonder what I would be giving up? I'd like to hear some
Jim Canfield wrote:
I've been considering replacing a PRI with SIP or IAX trunks. The
monthly cost difference is marginal, but it would save a bit on the
hardware side and soft trunks would be easier to manage. I can't help
but wonder what I would be giving up? I'd like to hear some
Jeremy Mann wrote:
Without knowing more, Why fix what isn't broken?
I should have stated, the PRI is on an existing PBX not asterisk. My
goal was to reuse the existing PBX PRI card to interface with asterisk.
I've been considering replacing a PRI with SIP or IAX trunks. The monthly
cost
Danger Wil Robinson!
Don't do it. If the cost is negligible, you are going to give up a huge
control / reliability factor. Unless you dedicated a T1 to just voice, you'll
not be able to guarantee quality.
I've had a few small companies use VOIP trunks with POTS backup, but I wouldn't
On Fri, 5 Oct 2007, Stephen Bosch wrote:
Here's what you'd be giving up: reliability.
It's basically an economic decision. If you're running voice over the
public Internet, you get what you pay for, plain and simple.
As others have suggested, dedicated IP over end-to-end loops to your
Steve Totaro wrote:
If your provider will be providing the SIP trunks then it might be OK.
Otherwise I would stick with PRI.
My lessons learned are that you cannot control the public internet and
traffic shaping and QoS are not useful most of the time. I would only
consider a point to
Get a 2 port card, problem solved. Asterisk is the Man-in-the-Middle.
I'm running this right now between an asterisk box and Nortel MICS system.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield
Sent: Friday, October 05, 2007 2:12 PM
To:
On Friday 05 October 2007 12:52:41 shadowym wrote:
I disagree with any argument for or against Digium in support of Asterisk
as much as I do for or against Sangoma or Rhino or one of the Chinese knock
offs in support of Asterisk. Digium uses the open source community to
create better
On Fri, Oct 05, 2007 at 01:26:58PM -0500, Lacy Moore wrote:
I was completely against the dual licensing in the beginning. But now, I'm
leaning more towards understanding it and the importance of it, especially
as it related to US Patent laws. We're going to find that everything is
patented
The distinction doesn't matter because in the end they can do what
ever they want with the code you disclaim to them. The whole thing
is very political and pointless to hash over and over again.
/b
On Oct 5, 2007, at 2:52 PM, Tilghman Lesher wrote:
When you contribute code to Asterisk,
On Fri, Oct 05, 2007 at 02:52:24PM -0500, Tilghman Lesher wrote:
On Friday 05 October 2007 12:52:41 shadowym wrote:
I disagree with any argument for or against Digium in support of Asterisk
as much as I do for or against Sangoma or Rhino or one of the Chinese knock
offs in support of
Because there is still old hardware in the pipeline at distributors and
resellers. They may even still be manufacturing some of that old hardware
so there is probably a lot of unrealized money involved. So in that sense I
can't blame them for being a bit hush hush about it's short comings.
Pepo wrote:
Hi friends.
How do I can forwarding a call to any extension in system A (running
Asterisk)
if it did not answear to the voice mail in the system B (other Asterisk)? so
later users of system A can check the remote messages.
Do i have to create each extension on B? Which are
Hi,
I have a really oddball time problem. When I check the server time using
'date' it is correct. When I review the time in Freepbx (under time
conditions) it is correct. When I look at the time stamp in the CDR it
is correct. When I review the time displayed for a voicemail in a web
browser
On Friday 05 October 2007 15:08:56 Brian West wrote:
On Oct 5, 2007, at 2:52 PM, Tilghman Lesher wrote:
When you contribute code to Asterisk, you retain ownership of your
code. You
are NOT disclaiming the contribution; you are LICENSING the
contribution.
This is an important legal
On Friday 05 October 2007 15:20:19 Tzafrir Cohen wrote:
On Fri, Oct 05, 2007 at 02:52:24PM -0500, Tilghman Lesher wrote:
On Friday 05 October 2007 12:52:41 shadowym wrote:
I disagree with any argument for or against Digium in support of
Asterisk as much as I do for or against Sangoma or
1 - 100 of 112 matches
Mail list logo
