It actually CAN but because someone was lazy and didn't want to
actually do the work to make it possible to do a full change during a
reload. The biggest issue is ztcfg would have to be absorbed into
chan_zap to make it 100% possible. In fact if Digium wanted to make
Asterisk easier to
Thats a great step forward. Auto for PRI doesn't make sense... but
two configs to describe the same thing makes no sense.
/b
On Oct 21, 2007, at 1:03 PM, Tzafrir Cohen wrote:
On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote:
It actually CAN but because someone was lazy and didn't
Make sure chan_zap.so is loaded.
/b
On Oct 18, 2007, at 9:34 AM, Pablo Almido wrote:
Hi List,
I am from Peru, I have installed an asterisk server in my company with
digium card E1 TE120P, I am having issues when i make calls, here the
error from my server
[Oct 18 09:13:50]
Why would a config error stop the module from loading? That seems
like a suboptimal behavior.
/b
On Oct 18, 2007, at 9:50 AM, Jared Smith wrote:
That would seem to indicate that the chan_zap.so module isn't being
loaded. What happens if you type module unload chan_zap.so and then
module
I'm sorry I call bullshit on this one. CentOS has been 2.6 for some
time.
/b
On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote:
Just 5 months ago CENTOS started to use Linux 2.6 one of the
reasons I'd abandoned for SuSE a while back.
You'll need to compile with debug symbols and have ulimited -c
unlimited set. Then you can examine the core and find out what
exactly caused the crash... Segfaults either are easy to find or very
hard to find, depending on what is happening. It could also be bad ram.
/b
On Oct 16, 2007,
You should really never touch those. If you're having problems with
the card call support because that is far from normal.
/b
On Oct 16, 2007, at 9:35 PM, Stephen Bosch wrote:
Eric Deutsch wrote:
Hi everyone, I’ve set up a little Asterisk system with a Digium
TDM400P
and everything
Just dont answer it till the processing is done. No debate is needed
for this. I do this millions of times per month.
/b
On Oct 11, 2007, at 2:56 PM, Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Victor wrote:
I need to process a number of lines of code in the dialplan before
Look at features.conf
/b
On Oct 10, 2007, at 10:48 AM, Reggie Payne wrote:
Hello All! I am new to the list. Does know how to record a call
on demand? What I would like to do is setup something that during
a call someone can hit a button a the call is recorded the after
the call is
On Oct 10, 2007, at 11:12 AM, Ex Vito wrote:
On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
zoachien wrote:
Google for mexuar.
Zoa
Or look at one that works with MS Windows, Linux or Apple
http://www.bicomsystems.com/products/C/P/319/382/
FYI, Mexuar's solution -- Corraleta
if you have allow=g729,ulaw and you want to use g729 but the current
channel is ulaw it will pick ulaw over g729 because it wants to
escape doing any transcoding if possible.
The best way to do this is setup different peers with different allow
lines to force the outbound leg to the codec
its IMS
/b
On Oct 9, 2007, at 10:39 AM, Andres wrote:
I had a friend yesterday showing me his new T-mobile blackberry with
WiFi Voip.I could not believe it until I actually saw him making
calls. There is no T-Mobile cell coverage at my house but he was able
to simply access the WiFi
http://www.imagestream.com/PCI_921-CDS.html
This card can do it. I have spoke with them about it and its very
capable of doing what is needed for a DS3 in a standard linux box.
/b
On Oct 9, 2007, at 10:42 AM, Andrew Kohlsmith wrote:
On Tuesday 09 October 2007 10:14:23 Matt wrote:
Before
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Baji
Panchumarti
Sent: Tuesday, October 09, 2007 12:07 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DS3 Interface
On 10/9/07, Brian West wrote:
http://www.imagestream.com/PCI_921-CDS.html
You apparently don't realize you're talking to. Thats ok, You keep
working on it from your angle. We are evaluating when the time is
right to implement this. We aren't doing this for Asterisk we are
doing it for FreeSWITCH.
/b
On Oct 9, 2007, at 12:00 PM, Steve Totaro wrote:
Well we are plugging it in the OpenZAP abstraction layer we have
already started on. This is usable by Asterisk also so asterisk
would benefit from it.
http://fisheye.freeswitch.org/browse/OpenZAP
/b
On Oct 9, 2007, at 12:31 PM, Steve Totaro wrote:
BTW, this is the wrong list if it not
And what was the purpose of this?
/b
On Oct 9, 2007, at 1:32 PM, Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, Brian West [EMAIL PROTECTED] wrote:
You apparently don't realize you're talking to. Thats ok, You
keep working on it from your angle. We
. were on their servers and
they called the FBI and three days later our office was raided. This
I consider mudslinging by you and wasn't very gentle man like.
/b
On Oct 9, 2007, at 1:32 PM, Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, Brian West [EMAIL
On Oct 9, 2007, at 2:12 PM, Matt wrote:
Perhaps it was uncalled for. However, if I were to consider using
FreeSwitch I would want to know who was/is behind it.
On 10/9/07, Brian West [EMAIL PROTECTED] wrote:
And what was the purpose of this?
/b
Matt, that was totally uncalled for.
Thanks,
Steve Totaro
Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
On 10/9/07, *Brian West* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You apparently don't realize you're talking to. Thats ok,
You keep
I would recommend doing it on a 64bit platform for sure. Not sure
Asterisk has very many linger issues on 64bit... I know I run it on
64bit without too much drama.
/b
On Oct 9, 2007, at 9:32 PM, Mr. James W. Laferriere wrote:
Please , step back form the keyboard , take a deep
Now the next question is why do no LGPL Dundi libs exist?
/b
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They why was it on the website?
/b
On Oct 8, 2007, at 11:59 AM, Tilghman Lesher wrote:
On Sunday 07 October 2007 15:23, Steve Totaro wrote:
How about the once announced Digium DS3 card (that I never saw
come to
market), that board must have some powerful onboard circuits or
require
a
The board never came to market 1. because the demand. 2. impossible
to do with zaptel.
/b
On Oct 7, 2007, at 3:23 PM, Steve Totaro wrote:
How about the once announced Digium DS3 card (that I never saw come to
market), that board must have some powerful onboard circuits or
require
a very
Telling someone to read the RFC bah.. might as well give them a
blanket and pillow because they will fall asleep. chan_sip is just
ugly in every way.
/b
On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote:
http://www.faqs.org/rfcs/rfc3261.html
as well as the source in asterisk (1.4.11
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
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
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.
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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
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
I think Lee Howard nailed it.
/b
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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,
I think the horse has been long dead!
/b
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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
___
In my opinion the dialplan isn't where that logic belongs.
/b
On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED]
wrote:
Hello,
I see that most people are using the extensions.conf syntax (most
of the
examples and questions here use that syntax). recently
6900
Fax: +27 (0)11 875 6901
Mobile: +27 (0)83 791 6662
Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Web:www.bitco.co.za
Brian West wrote:
In my opinion the dialplan isn't where that logic belongs.
/b
On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444
[EMAIL
On Oct 3, 2007, at 9:39 AM, Jon Schøpzinsky wrote:
Wouldnt that take a very large portion of datapower, to startup the
parsers and such, instead of having the whole dialplan natively in
Asterisk.
We always try to do as much as possible in dialplan, so that we are
not reliant on external
Its just a different way to express the same thing in a more fluid way.
/b
On Oct 3, 2007, at 10:33 AM, Anthony Francis wrote:
Doesn't this render having used AEL pointless?
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I'm growing fond of XML.
/b
On Oct 3, 2007, at 10:39 AM, Steve Totaro wrote:
To each his own. I like the flat files personally, they are more
fluid
to me.
Thanks,
Steve
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You still do not understand. It doesn't matter if the call coming in
is g729 you must transcode it to signed linear, mix the frames and
then code it back into g729 you end up with quality loss doing that.
/b
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a
Thanks for making it clearer :) My mind is mush today!
/b
On Oct 2, 2007, at 5:39 PM, Tilghman Lesher wrote:
Or, in other words, you cannot mix compressed data. You must first
decompress the data for mixing, then recompress it for transmission.
During both operations, there is a potential
Just buy the Linksys SPA962's they work better than the cisco phones
in a NAT env.
/b
On Oct 1, 2007, at 6:13 PM, Andrew Joakimsen wrote:
My understanding is:
Smartnet: service contract basically allows you to download the
newest sw release.
Besides that you can buy phones without a
Ok Let me chime in on this one.
If you can use ulaw/alaw because you'll end up with tandem encoding
which will make the conference sound worse to some people.
All audio coming in will get transcoded to signed linear and pushed
down into zaptel then back up and out to the conference
On Sep 28, 2007, at 4:52 PM, Mojo with Horan Company, LLC wrote:
To use the
wildcard characters, 'X', 'N', or '.', I had to also prefix my
extension with '_', which enables pattern matching.
Don't forget you also have Z which if I recall its 1-9, N is 2-9 and
X is 0-9
/b
Good luck with that one. Most unlimited providers have limits. (even
if they say unlimited)
/b
On Sep 19, 2007, at 12:32 AM, Jim Boykin wrote:
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
Their really isn't many differences. A true softswitch will usually
never speak to an end users device directly.
/b
On Sep 19, 2007, at 10:02 AM, satish patel wrote:
Dear all
what is softswitch what is difference between asterisk
and softswitch ??
regards
satish patel
Satish,
It depends on your goals. FreeSWITCH is approaching an official
release. Beta 1 is out now and various other tweaks in trunk. But
its really up to you to evaluate your need and compare which fits
your needs. I see them as complementary to each other so its really
up to you.
Asterisk isn't a big iron switch.
/b
On Sep 19, 2007, at 11:08 AM, Tzafrir Cohen wrote:
On Wed, Sep 19, 2007 at 11:15:25AM -0400, Alex Balashov wrote:
Asterisk is a PBX. A softswitch is more or less a fully featured
telephone switch, usually one that is extensively application-driven
(more
With zaptel that will be impossible, asterisk can do GR303 not sure
how well.
/b
On Sep 19, 2007, at 12:04 PM, Alex Balashov wrote:
Perhaps I'll be a little more amicable when someone finds a way to
bring
at least five or six DS3s into Asterisk.
It will not after some types of crashes.
/b
On Sep 5, 2007, at 9:43 AM, Perssy Llamosas wrote:
You are using safe_asterisk, it will restart automatically Asterisk
after it crashes.
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On Sep 5, 2007, at 7:42 PM, Philipp Kempgen wrote:
I don't need messages to tell me *5* times about Astricon,
who provides the bandwidth and how to unsubscribe.
You sure about that unsubscribe part? People do seem to miss it :P
/b
___
Sign up
Try setting the RTP packets to 0.020 instead of 0.030 which is the
default on the SPA's
/b
On Sep 3, 2007, at 5:00 PM, Todd Reese wrote:
Hi all,
I have just install and licensed Cepstral's Allison08kHz on my
Asterisk
1.4.11 system.
I can call the Allison's extension from my
Localnet is wrong... try localnet=192.168.1.0/24
/b
On Sep 3, 2007, at 9:13 PM, neoh kumyee wrote:
Hi,
I am trying to run an Asterisk (1.4.11) server on Linux Suse. The
server is behind NAT. I am testing with SIP client that developed
from PJSIP running on Pocket PC Windows Mobile 5.0 .
On Aug 30, 2007, at 8:49 AM, Matt wrote:
impressions are everything).Digium also makes money off of the
FXO/FXS/PRI cards, which you really wouldn't use unless you were
running asterisk. So in this case, while Asterisk IS free, it is
I have to comment here.
If I recall all the zap
On Aug 30, 2007, at 10:11 AM, Jared Smith wrote:
On Thu, 2007-08-30 at 15:42 +0100, Adrian Marsh wrote:
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
[local]
exten = _${SOMEVAR}.,1,NoOp(Dialled own number)
No, unfortunately you can't use variables
http://www.freeswitch.org/asterisk_stuff/app_distributor.c
/b
On Aug 30, 2007, at 7:38 PM, Paul Hales wrote:
We found the 'random' dialplan function worked quite well for
something
similar a while ago.
PaulH
On Thu, 2007-08-30 at 17:38 -0500, Carlos Chavez wrote:
I was wondering
On Aug 29, 2007, at 9:35 PM, Russell Bryant wrote:
Another Digium software developer, Joshua Colp, has recently been
working on an
automated build farm with virtual machines for all of the different
operating
systems we support. It already has 64 and 32 bit versions of Linux
(glibc and
On Aug 28, 2007, at 8:24 AM, Jody Gugelhupf wrote:
-- Now forwarding SIP/9083XXX-0816b208 to 'Local/
[EMAIL PROTECTED]' (thanks to
SIP/486-081d4738)
Because SIP/486 issued a 302 redirect to 247110358. Check the phone
for the forwarding setting.
/b
to improve our understanding of the dynamics of how everything works
together.
* Scaleability
* Reusability of code
* Standards (VoiceXML, MRCP and more)
If anyone is interested please email me off list and we'll plan on
having a meeting of minds.
Thanks,
Brian West
FreeSWITCH.org
On Aug 28, 2007, at 10:14 AM, Seysan wrote:
Hi all,
I'm kind a New to Asterisk.But I'm a Network Administrator with 5
years of experiance.
I want to know for an installation with 90 clients, If I don't want
to have just 1 server for it, then how is it possible to distribute
it
haha you going to be there?
/b
On Aug 28, 2007, at 9:30 AM, Chris Childress wrote:
oohs no!
Whats up, haven't heard much out of you lately.
Chris
Brian West wrote:
Everyone,
I will be attending Astricon in Phoenix and would like to have a
little get together
On Aug 28, 2007, at 3:49 PM, Doug Lytle wrote:
Christian Peter wrote:
Can anybody help me with this issue. Please no switch to Hylafax
mails, because I'm very happy with SpanDSP, it integrates nicely and
It just show you how many people on this list are pleased with
HylaFAX+
Doug
--
Having calls connected for that duration is worthless testing... What
you need to do is call setup and tear down many times per second... I
recommend trying to accomplish 20-30cps at 1ms to 10ms variable
durations. That will expose any bugs quickly.
And that my friend is how you expose any
This fails to take into account total failure of a machine. NAT
mappings and various other variables that are not covered by Dundi or
realtime... Best thing is to use OpenSER in the front then failure
isn't a huge issue.
/b
On Aug 28, 2007, at 4:40 PM, Bruce Reeves wrote:
Realtime and
On Aug 28, 2007, at 6:28 PM, Matt Riddell wrote:
Sorry to hijack the thread, but its great to see you here again Brian!
- --
Kind Regards,
Matt Riddell
Director
Thanks...
/b
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The HD Codec is just G.722
/b
On Aug 27, 2007, at 7:52 AM, Matthew Rubenstein wrote:
Do any softphones run the HD codec? What exactly is the HD codec
technically called, and is there any info about its codec running
inside
Asterisk?
On Mon, 2007-08-27 at 08:47 -0400, C F wrote:
The 601 has g722 (and its not g722.1 or .2)
/b
On Aug 27, 2007, at 8:14 AM, Bruce Reeves wrote:
The codec is G722 I believe. and Polycom has a conference speaker
phone with a subwoofer option that has HD voice.
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do any
FreeSWITCH supports 16k wideband conferences and supports G.722,
speex 16k and should work great with the phones that support it. I
have personally tested it with grandstream phones.
/b
On Aug 27, 2007, at 7:47 AM, C F wrote:
although not stereo i believe its the closest you will get if
If you can get an rtp debug while your pressing digits I can see if
maybe your device is sending the digits incorrectly.
/b
On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote:
my sip phone have dtmf relay: rfc2833
asterisk sip.conf have dtmf relay: rfc2833 in associated context.
I tried with
The way I said is the "gospel" of how it happens. /bOn Aug 16, 2005, at 1:42 AM, Erik Versaevel wrote:That should be controllable by a weight, for example 2 peers: ___
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Just an FYI http://www.groklaw.net/article.php?story=2005080914234645
/b
On Aug 16, 2005, at 4:50 AM, Tamas J wrote:
Joseph wrote:
I'll second that.
Hylafax has can handle the job. If you put asterisk in between
you are
looking for problems.
I've the following setup working with
Although Groklaw seems to think that these suits are about faxing, I don't think that they really are. See: http://www.hylafax.org/archive/2005-08/msg00107.htmlLee.No it is really about faxing. As someone that has first hand knowledge of the case outlined on Groklaw, it is in fact about
As someone that spent a week or more with anthm refactoring this code I can tell this is how it was when we were done and the code was accepted. So I do know a bit about this area of sip and iax./bOn Aug 16, 2005, at 3:03 PM, Pavel Jezek wrote:I remember many discussions about inteligent codecs
Time and time again. CHECK YOUR Span clock src./bOn Aug 16, 2005, at 10:18 PM, Ma Zhiyong wrote: Hi, I just setup a fax server by spandsp. But it doesn't look good. Because each fax I received from my fax machine is not completed. I use te410p work with it. While the voice call is good.
Here is an example:
Call comes in via PSTN... ulaw is the native format of the channel.
On the sip side you have g729,ulaw as the codec order. That call
will end up being ulaw because we send the native format as our first
choice above all because we don't want to transcode.
/b
On
You do realize that t.38 is the act of taking the t.30 stream and
stuffing into UDPTL packet and sending it over a network with a
little ASN.1 header added and some reliable delivery kinda like how
IAX has reliable delivery of UDP packets used for signaling. This is
a very basic
The TNT can't pass callerid name as far as I know./bOn Aug 9, 2005, at 5:17 PM, Damon Estep wrote: Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? ___Asterisk-Users
What are the advantages of using woomera IAX2 instead of native IAX2?Put woomera aside right now, This is something that brings a cross platform IAX2 stack that can for example be used in Gnomemeeting or anything else that uses OPAL, using a closed and open familiar API. This can be used on
Minessale II to interconnect your asterisk systems and use the IAX2,
SIP, and H.323 protocols.
I would like to thank everyone involved in Cluecon for all their
support!
Thanks guys!
Brian West
Asterlink.com
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Its very clear your zaptel.conf and/or zapata.conf is wrong.Make sure your devices are registered.. re-run ztcfg -vvv/bOn Jul 28, 2005, at 7:48 AM, Dr. Marios Moutzouris wrote: Hello, This is debug output I get: Jul 28 15:05:49 WARNING[8249]: chan_oss.c:239 sound_thread: Read error on sounddevice:
environment.
I would like to personally thank Mark Spencer and Digium for their
support.
Thanks,
Brian West
Asterlink.com
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for something else thus
causing the L option to fail/reset the timer to zero thus causing it
to never timeout if someone were to say press a DTMF digit.
So if you use this please test this and report back to the bug ASAP.
Thanks,
Brian West
If you use mp3nb from the sample configs you will have exactly 1 per
class.
/b
On Jul 26, 2005, at 9:38 PM, MF Hulber wrote:
Yes, I always have two.
MARK.
Billy Dunn wrote:
Does everyone have two processes running for mpg123? I always
have them when I'm running an idle Asterisk box.
http://www.globalipsound.com
Try there.
/b
On Jul 25, 2005, at 8:15 AM, Eric Wieling aka ManxPower wrote:
Steve Underwood wrote:
Steve Kennedy wrote:
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen
wrote:
I don#8217;t know if I have the same experiences. Usually my
Cluecon's ( http://www.cluecon.com ) premier sponsor, Sangoma ( http://www.sangoma.com ), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma’s next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your
I'm going to be speaking about how to use valgrind, gdb and strace to
help debug issues... it can be applied to more than just asterisk.
/b
On Jul 25, 2005, at 10:29 AM, Terry Moore-Read wrote:
I'm relatively new to Asterisk and I'm hoping attending Cluecon
will be
a good way to get up to
TDM400P 4 port analog card. (winner picks configuration)
2 Pre-Paid Asterlink accounts with 1000 minutes of talk time.
Tickets will be issued with your ID badge at registration. The
drawing will take place at Noon each day right before we break for
lunch. Good Luck!
Thanks,
Brian West
Cluecon's ( http://www.cluecon.com ) premier sponsor, Sangoma ( http://www.sangoma.com ), will be giving away 3 A101 single-port T1 cards during Cluecon. The A101 is Sangoma’s next generation hardware designed for optimum support of data and voice over T1 and E1. Register for Cluecon now for your
I'm going to be speaking about how to use valgrind, gdb and strace to
help debug issues... it can be applied to more than just asterisk.
/b
On Jul 25, 2005, at 10:29 AM, Terry Moore-Read wrote:
I'm relatively new to Asterisk and I'm hoping attending Cluecon
will be
a good way to get up to
I'll talk to your boss if he has a problem! ;)
/b
On Jul 23, 2005, at 11:03 PM, Terry Moore-Read wrote:
Mine did.
[EMAIL PROTECTED] 7/21/2005 2:54 PM
Brian West wrote:
ClueCon is coming in 2 weeks so we urge everyone who plans on
attending to register today so we get a proper
PLEASE FOR THE LOVE OF GOD put a NAME in your email program.. I'm
sure it makes going back and finding stuff in the archives when you
and about 100 other people use Asterisk in their names This
goes for anyone that uses Asterisk, Asterisk PBX or any form
there of .. lets put a name in
But I guess I'm wondering ... does the present licensing model discourage other vendors from contributing to *? I'm not sure Sangoma developers could sign the disclaimers even if they wanted to ... but then again I don't know if there's anyone there with anything to offer. I would think that that
Aidan isn't a troll he does raise a very valid point.
/b
On Jul 23, 2005, at 5:55 PM, Brian Capouch wrote:
Aidan Van Dyk wrote:
Is this indicative to how Digium people respond to everything
(including the
company that built the first asterisk-supporting hardware still
continuing
to
Or better yet.. modify the disclaimer like I and a few others did to
say that the only thing you will disclaim are things you post on the
bug tracker! NO UPDATES, NO CHANGES, NO NOTHING! If its not posted
under your user on mantis IT IS NOT DISCLAIMED!
/b
On Jul 23, 2005, at 2:59 PM,
I'm shocked nobody put the new charlie and the chocolate factory
soundtrack on the list...
/b
On Jul 22, 2005, at 9:57 AM, Michael Graves wrote:
I was thing about XTCs stupidly happy
M.
On Fri, 22 Jul 2005 15:57:07 +0200, Simone Cittadini wrote:
Happy Tree Friends' theme is all you
that are freely
available:
* res_perl - Embedding Perl into Asterisk
* res_js - Embedding JavaScript into Asterisk
All of this for the modest cost of $350.00. You could learn enough
the first day to justify the price and then you get 2 more days on top
of that!
Thanks,
Brian West
Asterlink.com
Kristian Kielhofner, the lead developer of the AstLinux project, will
be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a
complete Asterisk distribution built to run from Compact Flash and
uses less than 32mb.
Thanks,
Brian
___
Do you even know what e.164 is?
http://www.numberingplans.com/index.php?goto=guidetopic=E164
/b
On Wednesday, July 13, 2005, at 09:27PM, Julian J. M. [EMAIL PROTECTED] wrote:
Have you tried googling for asterisk e164 ?
Julian.
On 7/13/05, Will Velez [EMAIL PROTECTED] wrote:
Hi my name is
to reserve you a room!
Thanks,
Brian West
http://www.cluecon.com
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