Hi,
My question is more what should be done than how should it be done.
I could say :
If you were a teacher, teaching and preparing your courses once a week (as
you can't be called while teaching, can you ?) would you prefer your phone
system to log you in or out
1- automatically according a
EDIT:
It seems that if the call fails then I can see the
number in the cdr-lastdata (and lastapp) fields (eg.
Dial Zap/g1/7032).
However, if the call is answered, there's no trace of
the number if Zap was used (there's a trace only if
SIP is used).
So, how can I hack this so that I can set the
Hi,
It is possible to jump into a Macro (or some similar dialplan jump)
when a transfer causes a call to be re-bridged? I do not believe that
GOTO_ON_BLINDXFER will do the job, because we use SIP phones, and use
the handset's own transfer or blind-transfer facilities.
What I want to achieve is
On Fri, Aug 10, 2007 at 07:52:30AM -0400, Steve Totaro wrote:
Sure. But we were talking about installers who do it *wrong*.
-- jr 'at least, *I* was' a
Luckily, I was trained by a guy that had been doing telcom work for
forty years. He used to be a lineman in the Philippines (no bucket
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single
On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a
Mike wrote:
The thing is that I make them automagically reload from outside Asterisk (by
calling asterisk -rx extensions reload)
Mike
Why would you do that? There is no real point in reloading configs
unless they have changed.
Anthony
___
Mike wrote:
The thing is that I make them automagically reload from outside Asterisk (by
calling asterisk -rx extensions reload)
Correct me if I am wrong, but can't you load and unload individual
extensions from the console, or through the AMI?
That's what I meant you can script this.
Andrew Kohlsmith wrote:
On Thursday 09 August 2007 1:18:17 pm Stephen Bosch wrote:
Why would anyone want a Collingwood DID? I don't answer calls from
Collingwood simply because I am plain old not interested in the free
vacation weekend I keep winning. :-)
Are there lots of boiler rooms in
Jay R. Ashworth wrote:
On Fri, Aug 10, 2007 at 07:52:30AM -0400, Steve Totaro wrote:
Sure. But we were talking about installers who do it *wrong*.
-- jr 'at least, *I* was' a
Luckily, I was trained by a guy that had been doing telcom work for
forty years. He used to be a lineman in the
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single phone and then try to enable monitor, when
I pick up the
Hello all,
After installing Asterisk, i have installed the docs by make progdocs.
But i don't know where to locate this documentation.
please Help.
Thanks.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I want to break it into pieces so that if dave leaves, we can
just record that
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
The other issue is scale. A single Class 5 switch shelf can take many
DS3s, and even several OC-Xs.
Yeah.
phew
Asterisk can take... what, a few T1s? Maybe? That's nice. And entirely
worthless. Someone is going to have to
I would like to setup my fax extension through freepbx to NOT have to
dial 9. I will never dial internal numbers, so all I want it to do is
pass the digits to the trunk. Is that possible with freepbx and if so,
how is it accomplished?
Thanks in advance
Sean Garland, V.P.
Siskiyou Technology
. . .
The cost of the wire is not that much, without even shopping around or
going through my regular distributor I found this link.
http://www.wesbellwireandcable.com/Bare_Tinned_Copper.html?gclid=CNLa25jj6Y0
CFQ1zHgodBychsQ
1000ft = $169.
Again, it is not that big of a deal if
On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
That's great, now say you have 5 or 6 AA's and each one has 10 different
parts that you want to record (thank you for calling... for Steve
press 1 for dave press 2). Rather than having to record a long
message, I want to break it
Mike wrote:
Well, if you really must know (this is OT for everybody else I guess) I have
a custom Web GUI used for my customers, and when some settings are modified,
a conf file is created. This conf file must be reloaded at this point,
therefore I call the reload command externally.
Why do
Gordon Henderson wrote:
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it.
I can't help but think you're making life hard for yourself.
Why not do it by
Strangely enough i ordered another number on their website yesterday hoping
that they might have filtered numbers which are to be disconnected from the
pool but after a hour of registration i get the same email for new DID i
bought . This is pathetic and frustrating now . I have sent an e-mail to
The other issue is scale. A single Class 5 switch shelf can take many
DS3s, and even several OC-Xs.
Asterisk can take... what, a few T1s? Maybe? That's nice. And entirely
worthless. Someone is going to have to make it possible to take at least
a few DS3s in a PC before something like
On Fri, 10 Aug 2007, Cesc Santa wrote:
inline ...
On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 10 Aug 2007, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
Installed from binary or compiled by yourself?
I compiled it myself ...
OK, Great.
I just took a peak at the
On Fri, Aug 10, 2007 at 09:42:39AM -0500, Todd Adamson wrote:
I know there are differences between a PBX switch and a CO switch.
Yeah, kinda.
Some colleges use ATT 5ESS-2000s as PBXen.
Can Asterisk completely replace and act as a CO switch? Are there any
telecoms out there using Asterisk
Todd Adamson wrote:
As I am working my way to understand Asterisk, I have a couple of
questions that hopefully someone will answer.
I know there are differences between a PBX switch and a CO switch.
Can Asterisk completely replace and act as a CO switch? Are there any
telecoms out there
inline ...
On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 10 Aug 2007, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
Installed from binary or compiled by yourself?
I compiled it myself ...
I just took a peak at the command: show translation
and I saw that I can
On Fri, Aug 10, 2007 at 02:04:45PM -0400, Alex Balashov wrote:
Asterisk is _NOT_ a switch. Asterisk is not a transit element.
Asterisk is an *endpoint*. It makes for a nice PBX, feature server, etc.
Kind of like the BroadSoft, but on a much smaller scale.
I hadn't been tracking oSS7
I am having a bit of a problem implementing the pickup command in my
dial plan. I have setup this rule:
exten = _*8XXX,1,Pickup(${EXTEN:2})
This works as expected when someone dials an extensions number and I
can get the call. The problem I have is that when a call enters my
Alex Balashov wrote:
On Fri, 10 Aug 2007, Jay R. Ashworth wrote:
Short version: There's some hope Asterisk could handle the programming,
but the switching fabric simply is *not* up to the task yet.
And I am not sure that kind of DSP density or CPU-bound framing and
transcoding is even
On Fri, Aug 10, 2007 at 07:07:35PM +0200, MOSBAH ABDELKADER wrote:
Hello all,
After installing Asterisk, i have installed the docs by make progdocs.
This produces docuemntation under docs/ . Id does not install them.
But i don't know where to locate this documentation.
In the source
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
___
--Bandwidth and Colocation Provided by
On Fri, Aug 10, 2007 at 12:38:09PM -0600, Stephen Bosch wrote:
I read an article about a Luftwaffe pilot who broke the sound barrier in
an Me262 (the WWII jet fighter). Of course, he did it in a dive. The
claim was disputed.
An aeronautical engineer was quoted as saying, Even if it were
Hello,
I am trying to create a Java GUI that will interact with an Asterisk Server.
This Java GUI will essentially be a custom made SIP softphone. I will most
likely use the Asterisk-Java Live API to create the connection to the Asterisk
server and to open a new call. Then, I plan to use the
On Fri, Aug 10, 2007 at 09:50:49AM -0600, Stephen Bosch wrote:
Now, I can go into any telco closet and know quite a bit about the
installer's ability and work ethic.
Yep.
Course, some of them aren't up to it anymore, though I did see a CLEC
installer do a Bell-quality job a couple
The thing is that I make them automagically reload from outside Asterisk (by
calling asterisk -rx extensions reload)
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Friday, August 10, 2007 10:32
To: Asterisk Users Mailing List
Jay R. Ashworth wrote:
On Fri, Aug 10, 2007 at 09:50:49AM -0600, Stephen Bosch wrote:
Now, I can go into any telco closet and know quite a bit about the
installer's ability and work ethic.
Yep.
Course, some of them aren't up to it anymore, though I did see a CLEC
installer do a
As I am working my way to understand Asterisk, I have a couple of
questions that hopefully someone will answer.
I know there are differences between a PBX switch and a CO switch.
Can Asterisk completely replace and act as a CO switch? Are there any
telecoms out there using Asterisk as a CO
I don't know how to keep the MyStatus and Buddies from showing up when
presence
is turned on, but if it helps, you only need to turn it on for phones that
NEED to see
the OTHER peoples presense. For example, turn presence on for your
secretaries
phones, but not on for the bosses. At least the
On Fri, Aug 10, 2007 at 11:35:32AM +0200, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
I just took a peak at the command: show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do I need a license or compile something extra?
The G723,
Jay R. Ashworth wrote:
On Thu, Aug 09, 2007 at 09:22:34PM -0400, Steve Totaro wrote:
I try not to spring any surprises on my customers. If it is a new
punch-out, I will make sure that building maintenance is aware of the
requirements. They are usually very helpful for these kind of
Ken,
You understood correctly.
For those who answered and didn't understand the need, I wanted to reload
automatically (based on some external event) part of my conf file. I only
felt that since this was automatic, it would have been better to limit this
reloading to only the part that
Mauro Zanin wrote:
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron inside. The PC seems smart
enough an rapidly the CentOs and Asterisk were loaded on it. I has only 2
extensions(SIP telephones, one GPX2000 and one Grandstream 486) and one ISDN
adapter with Bristuff. Web server
On Wed, 2007-08-08 at 22:30 +0800, Steve Underwood wrote:
Patrick wrote:
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
How can one verify that IAX trunking is in effect and that Asterisk is
trunking multiple call paths between two Asterisk servers?
With 1.4.10 on both ends, entering iax2 set debug trunk on either end
merely results in the response:
IAX2 Trunk Debug Requested
and nothing more.
--
George
On Fri, 10 Aug 2007, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
Installed from binary or compiled by yourself?
I just took a peak at the command: show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do I need a license or compile
Hi,
I have asterisk 1.2.18.
I just took a peak at the command: show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do I need a license or compile something extra?
The G723, 726 and 729 ... I need a license, is that it? one for all of them?
Steve Davies wrote:
On 8/9/07, Oscar Patricio [EMAIL PROTECTED] wrote:
Hello!
I have following scenario:
PBX - Asterisk - ISDN E1 line
The asterisk box relays calls from the E1 to the PBX and vice versa.
Additionally some outgoing calls of the PBX are being sent over VoIP
providers
Hello Gordon,
Thursday, August 9, 2007, 4:39:44 PM, you wrote:
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
No that doesn't work. You can't dial this number. You have to send
special facility keypads to telco switch. Normal
MOSBAH ABDELKADER wrote:
Hello,
Is the OpenVPN the ideal solution to set a tunnel between two asterisk
servers or there is a better solution.
Mosbah,
We use OpenVPN between 3 facilities and Asterisk between them. We have
OpenVPN on their own systems and Asterisk sets behind them.
I don't
Well, if you really must know (this is OT for everybody else I guess) I have
a custom Web GUI used for my customers, and when some settings are modified,
a conf file is created. This conf file must be reloaded at this point,
therefore I call the reload command externally.
Why do I do this?
I have never thought about it, but you may want to have a look at some
http unit-test framework - they usually provide proxy services that are
able to automate and script a generic http conversation.
l.
In data Fri, 10 Aug 2007 12:13:17 +0200, Olivier [EMAIL PROTECTED] ha
scritto:
hello,
If your phones are incapable of receiving calls, I would guess that they
would also be incapable of receiving anything else from the SIP/Asterisk
server that would force a re-register.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
OpenVPN is very good in NAT (if one of your boxes is behind NAT). otherwise,
OpenVPN seems to be a bad choice, it's complicated, non-standard (there'n no
RFC on OpenVPN).
On 8/10/07, MOSBAH ABDELKADER [EMAIL PROTECTED] wrote:
Hello,
Is the OpenVPN the ideal solution to set a tunnel between
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron inside. The PC seems smart
enough an rapidly the CentOs and Asterisk were loaded on it. I has only 2
extensions(SIP telephones, one GPX2000 and one Grandstream 486) and one ISDN
adapter with Bristuff. Web server has about a delay of 20
hello,
I would to define and unattended process to configure devices which are
http-server-enabled, use DHCP but do not use TFTP-DCHP to configure
themselves during boot.
Has anyone worked on such subject ?
I was thinking of something like :
populating configuration file from device web pages
Hi Mauro,
changing from pmtp to ptp port type increased the time to call outside.
Log files show the card waiting for something from the telco line.
What kind of provider and ISDN port type have you got?
Giorgio
Mauro Zanin wrote:
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron
OK, now I know for sure... This is roughly 20 minutes after
the 601 crashed... There is an abandoned 'meetme' hanging
around in asterisk as seen below.
Conf Num PartiesMarked Activity Creation
1913938683d0006 0001 00:19:07 Dynamic
* Total number of MeetMe
On Thu, Aug 09, 2007 at 08:12:12PM -0500, David Bandel wrote:
Folks,
I'm trying to implement a simple loop in a dialplan. The object is to
set a counter, run through some IVR options, increment the counter,
return to the start, then finally fall through to an operator or
voicemail.
Am
On Fri, 10 Aug 2007, Jay R. Ashworth wrote:
Short version: There's some hope Asterisk could handle the programming,
but the switching fabric simply is *not* up to the task yet.
And I am not sure that kind of DSP density or CPU-bound framing and
transcoding is even possible. At the very
Mike wrote:
Ken,
You understood correctly.
For those who answered and didn't understand the need, I wanted to
reload automatically (based on some external event) part of my conf
file. I only felt that since this was automatic, it would have been
better to limit this reloading to
Hello everyone,
I'm hoping someone can help me with this. I have a business customer in
the U.S. (Michigan, ATT Territory).
I need to get 4 trunks into an asterisk Box. My intention is to use an
Eicon Diva Server card with 2 BRI Circuits. The reason for this is that
the business needs DID's
On Thursday 09 August 2007 1:18:17 pm Stephen Bosch wrote:
Why would anyone want a Collingwood DID? I don't answer calls from
Collingwood simply because I am plain old not interested in the free
vacation weekend I keep winning. :-)
Are there lots of boiler rooms in Collingwood?
...
When I ordered BRI in Michigan (25 miles N. of Detroit) It was still SBC.
I also had a hard time finding the right dept.
I do not know if it changed since then, but I had to order BRI from the same
department that sells P2P T1s, and DSL.
Even though it was a voice service, the data group sold
On Fri, 10 Aug 2007, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
He said _switch_ TDM. :-)
There's
Hi!
At the moment i am using a digium tdm400 card for my analog phone lines.
The zaptel driver supports fax detection, so incoming faxes are
redirected to the fax extension automatically.
This works without problems with asterisk 1.2.
But now I would like to switch to ISDN (mISDN) and asterisk
On Fri, 10 Aug 2007, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
And BT's 21cn (21st Century Network) is touted
On Fri, 10 Aug 2007, Gordon Henderson wrote:
On Fri, 10 Aug 2007, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really? http://www.pt.com/products/prod_segway_ntwksolution.html
On Thu, Aug 09, 2007 at 07:53:07PM -0600, Stephen Bosch wrote:
Mike wrote:
In the interest of making things cleaner, I'd like to know if I can just
reload one single conf file. Let's say I have two files, extensions.conf
which includes small_file.conf.
I only want small_file.conf
It wasn't that long ago that repeated reloads would cause an Asterisk
system to slow and eventually crash. I think that bug was fixed but I
would still only use reload on specific modules, not a total reload.
Thanks,
Steve
Mike wrote:
Ken,
You understood correctly.
For those who
On Fri, 10 Aug 2007, Olivier wrote:
hello,
I would to define and unattended process to configure devices which are
http-server-enabled, use DHCP but do not use TFTP-DCHP to configure
themselves during boot.
Has anyone worked on such subject ?
I was thinking of something like :
OpenVPN is a great choice. It is it's own standard, it does not need a
RFC, and it is not really complicated at all.
Find a good OpenVPN howto (skip the manual for now). It will take you
step by step on how to setup a tunnel. Once you have your tunnel up,
start reading the manual (if you
Hi,
I just enable presence on my phones, so that secretaries can see whether
their bosses are already on a call or not.
Unfortunately, this came with the added benefit of adding two soft key
features on the screen, MyStatus and Buddies, which I want removed (it's
good enough for me that they
Yep I run 2ghz Celeron for my home system and very responsive.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve
Watch the mpg123 process, it can take 99% of available cycles and slow
everything down.
I would disable http, cups, smb and any other non-vital process, reboot
and see if things are better.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305)
Mauro Zanin wrote:
Hi everybody,
I installed a 3.0Gb 512MB TrixBox with a Celeron inside. The PC seems smart
enough an rapidly the CentOs and Asterisk were loaded on it. I has only 2
extensions(SIP telephones, one GPX2000 and one Grandstream 486) and one ISDN
adapter with Bristuff. Web server
run iax2 show peers and see next to port (T) if it comes then you are using
IAX2 Trunking feature.
On 8/10/07, George Pajari [EMAIL PROTECTED] wrote:
How can one verify that IAX trunking is in effect and that Asterisk is
trunking multiple call paths between two Asterisk servers?
With 1.4.10
Alex Balashov wrote:
On Fri, 10 Aug 2007, Gordon Henderson wrote:
On Fri, 10 Aug 2007, Anthony Francis wrote:
On Fri, Aug 10, 2007 at 11:37:37AM -0400, Alex Balashov wrote:
And as a CO switch, you *must* switch TDM; VoIP isn't really an option.
Really?
Just one question, why would the Asterisk be involved in the voice path
at all ?
I would assume a media gateway (TNT ?) would be the obvious choice to
provide trunking side. And, for line side another gateway (not so sure
would be as often seen), but in this case a Line side gateway, and
Too bad, I think those two items on the menu confuses my (easily confused)
users.
As for changing the string, you can do that yourself by correcting the
language files if you use FTP or TFTP provisioning. I have to do this often
with the French strings, with are REALLY badly translated and full
inline
On 8/10/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Fri, 10 Aug 2007, Cesc Santa wrote:
I can use them ... but cannot do translation ... only pass-through ...
it
complains that
there is no function to convert to internal codec (pcm?)
G726 should also be compiled in as
Hi:
Is anybody out there successfully using Mitel SIP sets with Asterisk? I
hear they're not the most standards-friendly, and don't play well with
non-Mitel switches.
I have a pile of them and would like to see if I can use them, but not
if it promises to be a hassle.
-Stephen-
I've found OpenVPN to be easy to configure and very robust. It has a
zillion options, but they are just that - options. I haven't used it for
VoIP, but I've put it to good use doing layer 2 bridging which has
eliminated many problems with certain programs traversing NAT and
load-balancing
I usually have good results when using a regular fax machine connected
to a PAP2T on a small network. I have a customer that has this setup in
several offices. Lately I have noticed that recent versions of Asterisk
have worse results with this fax setup that onlder versions. I have 3
Hi there,
Could everyone that has a working production Asterisk server that uses a
Digium telephony card as a BRI/PRI gateway let me know what
motherboard/processor your server uses?
We use a Digium TE210P as our telephony card. We have tried a couple
motherboards, and neither is giving us
On 8/10/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Aug 10, 2007 at 11:35:32AM +0200, Cesc Santa wrote:
Hi,
I have asterisk 1.2.18.
I just took a peak at the command: show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do
C F wrote:
This is the postmaster at the list and I am notifying you that your
message failed.
Over the past two days my new posts seem to have silently been dropped.
I wonder if I can reply to an existing thread...
--
Does your Canadian VoIP service need CRTC-compliant 9-1-1 services?
Perhaps. I'm interested in knowing what this is all about. Hopefully
it's just Fonality trying to create a new revenue stream, kinda like
Digium did w/ ABE. I'd hate to see them dump the open part of their
community. That community is very valuable for beta testing, giving
feedback,
I was very shocked when I first heard that Fonality purchased TrixBox,
it didn't really make business sense to me. I still cannot see the sense
unless you are correct in the ABE comparison.
Not sure what all the licensing in TrixBox is but if they dump the
open, can't we always just fork. I
I don't know what ever happened to the DS3 card that Digium was supposed
to release. Maybe it was just for media hype or maybe the card had too
many issues to be released. Maybe we will see it one day.
http://www.voip-info.org/wiki/view/Digium+DS3000P
Thanks,
Steve
Alex Balashov wrote:
The
Not to SPAM their unveiling, but I do enjoy reading the trixbox forums as
well, and here was part of a message that Kerry posted about the new product.
-quoting kerryg-
trixbox Pro is NOT simply a different support/release model for CE like many
other open source projects do, trixbox Pro is
Why don't they say FreePBX. After all trixbox is all about FreePBX. If they
remove FreePBX from Trixbox, nothing is left in it. A half working HUD, and
another small little things don't make any major difference after all.
So are the FreePBX developers with the Fonality team or with the open
On Fri, 10 Aug 2007, Steve Totaro wrote:
I don't know what ever happened to the DS3 card that Digium was supposed
to release. Maybe it was just for media hype or maybe the card had too
many issues to be released. Maybe we will see it one day.
If it ever happens, it will certainly take
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