On Thu, 2006-06-01 at 21:16 -0700, Mike Fedyk wrote:
The way asterisk works is it abstracts concepts from protocol details.
sorta, it would be better if it actually abstracted everything so that
applications (modules prefixed with app_ generally) dont have to know
much, if anything, about a
this should help You
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+system
Is there away to set in the dialplan to have asterisk send an email to
someone letting them know a caller called with Timedate caller Id and so
forth.
My system is set to call in say extension 100 goes to
Hello,
I have an issue with the IP Phones Grandstream BT101/102. At a random
time (more than one hour of inactivity) the phone unregister from
Asterisk and became unavailable for incoming calls. The user can not see
nothing in the LCD of phone and the calls can not ringing...
Anyone has look
Hello Masters,
I am using SER as proxy and registrar and redirect server and iam using
MEDIAPROXY for SER to handle Nated calls ,
So, now i want to connect Asterisk to SER to handle only pstn calls so
how the process here goes on ... I mean is, when a sip user who
registerd at SER calls to pstn
On Jun 1, 2006, at 10:57 PM, Dan Austin wrote:
It was put in a development branch, but has not seen any action in
over two months.
I have had it running against chan_ooh323 for six months and chan_sip
for three weeks in production for a moderately loaded conferencing
server. Absolutely no
Well,I
decided to go with the Aastra 480i CTs. I should have them in a couple
days, so I'll let you all know what I think. Thank you all for the
discussion and information that helped me make my decision.
Regards,
George A. Roberts IV
President and CEO, Interjuncture Corp.
We will be closed in observance of the Shavu'ot holiday on Friday, June 2. We
will respond to any messages on Monday morning. Thank you for contacting
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ViaTalk shows that they have service in Houston and they support porting.
http://www.viatalk.com/
Been using them for about 6 months with our * box and they've been rock
solid.
Regards,
George A. Roberts IV
President and CEO, Interjuncture Corp.
http://www.interjuncture.com/
-Original
- Original Message -
From:
Eberhard Müller
To: asterisk-users@lists.digium.com
Sent: Thursday, June 01, 2006 7:58
PM
Subject: [Asterisk-Users] Re: Bristuffed
Asterisk: Hangup problems
Hi,
i have nearly the same problem. But I took the
lastest
I had a similar problem with the GXP-2000 phones and firmware 1.0.1.9.
Upgrading to the 1.0.2 (now running 1.1) firmware fixed it.
On Fri, 2006-06-02 at 08:37, Información Capa Tres S.L. wrote:
Hello,
I have an issue with the IP Phones Grandstream BT101/102. At a random
time (more than one
I am trying to get to the bottom of audio clicks, pops, dropouts with my
Asterisk server. These occur even when the system is under minimal load
(e.g. 1 Zap device in a queue being played music on hold) and occurs with
both Zap and Sip devices so isn't network related. The audio problems occur
Mike Hammett wrote:
I'm looking for an ATA\Voice Gateway that runs IAX and has several
ports (8 would be nice). I am looking to avoid devices that use the
same firmware as the ATCOM devices as I found them to be buggy (and a
PITA to find the proper update).
The Atcom devices use 2
Hello,
At a client site yesterday I installed a dozen GrandStream GXP-2000's
with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX
and phones: network access for users windoze PC's through the phone's
switch port was unbearably slow, making it almost impossible to work.
Hi all,
I am trying to find a way to set the maximum wait time for a user in MeetMe.
I am creating a dynamic conference.
If there's only 1 user in the conference room, I would like to allow him
to wait only until the certain time. eg. may 1 min.
If there's no other user joining the conference,
With all this weather stuff going on lately, I also created a small
howto for using with cepstral.
Hope someone makes use of it:
http://www.voipphreak.ca/archives/269-Even-More-Asterisk-Weather-Now-Cepstral.html
Thanks,
Matt G
www.voipphreak.ca
www.asterisk-jobs.com
Kerry, so to park a call, you would put the line you are on on hold, hit line 2, dial 700 (or whatever your park ext is) listen to find out the number, then hit TRNF and hit line 1.
That's a lot of work to park a call. I just realized this might be a problem. I'm about to put 4 phones in an open
Vic wrote:
Hi, I was wondering if anyone knows of a opensource SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP packets to it, I would be
able to do it. The problem is: has
Muchas gracias Felix, voy a probar a ver que tal
jala.
Tu tuviste ese miusmo problema?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martinez
FelixSent: Thursday, June 01, 2006 9:28 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
On Fri, 2006-06-02 at 12:01 +0200, Louis-David Mitterrand wrote:
Hello,
At a client site yesterday I installed a dozen GrandStream GXP-2000's
with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX
and phones: network access for users windoze PC's through the phone's
Sean,
Where did you find that quote, I would like to see the rest of the
thread if there was relevant discussions.
Thanks.
It was really a two email thread... I had sent an email asking what the
status of BLA/SCA: Here is the entire thread:
Sean Cook wrote:
I take it SCA/BLA isn't
I don't really see what you mean with So an in my q931 it is changed to
ourcallstate.
Did you apply the proposed patch yourself? Because it isn't implemented in
any of the official Bristuff tarballs afaik.
So did you untar the Bristuff, downloaded the asterisk libpri sources, and
applied
Lacy,I am in a similar situation, except that my users are extensions aware. However, I'd love to know how you solved your problem since call transfer seems a bit complicated at the moment.Thanks,DanielOn Jun 2, 2006, at 6:51 AM, Lacy Moore - Aspendora wrote:Kerry, so to park a call, you would put
Here ya go:
For configuring the speakerphone volume after a phone restart
(default volume) you need to set in sip.cfg:
voice.gain.rx.digital.chassis="X" X been the default volume -15,
-9 ... 0... 9...15 in 3 intervals
For configuring the ring volume after a phone restart (default
volume)
FYI, I was having problems getting chan_misdn to work, it just wouldn't
get the extension in immediate mode. chan_capi got the extension okay
but the audio quality was awful.
In the end, I put a Wait(0.01) before Answer in the incoming mISDN
context, then DISA(no-password|sip_provider_out) and
I've been using Viatalk for 2 months and have been down 3x. They had
a misconfiguration on there end that they would not fix until quite a
few of us started having the issue. Tech support has been pretty bad
for the most part and unwilling to cooperate. So I would not suggest
them at this
2006/6/1, Attilla De Groot [EMAIL PROTECTED]:
Well I just found this:
http://bugs.digium.com/view.php?id=5162
So it seems that there is a patch and that it's ready for 2 months,
but I just checked the rtc.c code and it doesn't include this patch.
And I don't like to use beta patches on a
2006/6/2, Attilla De Groot [EMAIL PROTECTED]:
But to be honest, I'm not really a programmer and I'm not sure how to
apply this patch on the source code. Could you give any instuctions
about how to apply the patch ?
To apply a .patch file, you should use patch -pn, but if your source
code is
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Thursday, June 01, 2006 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] addons trunk make error
There are too
Hi
All,
Somebody here has
experiences with asterisk server which trunks to a cisco 2851 via
sip/h323.
The cisco is the
gatekeeper to the pstn network.
Somebody has a
sample configuration here for the cisco?
Regards
rene
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si, al principio, de ahora en adelante en todas la instalaciones qe hago codificamos ese parametroOn 6/2/06, Anton Krall
[EMAIL PROTECTED] wrote:
Muchas gracias Felix, voy a probar a ver que tal
jala.
Tu tuviste ese miusmo problema?
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Yes, their name has come ujp on the list enough to make me wary. I'd rather use someone who's primarily a wholesaler if possible. Not that I push that many minutes, but I don't mind paying for quality support.
I've been trialing CarolinaNet (a Nuvio reseller) and the service has been good.
Pues muchas gracias por el tip.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martinez
FelixSent: Friday, June 02, 2006 8:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Unicall Protocol Failure
si, al
Hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
hi,
maybe http://www.oreka.org
--- Vic [EMAIL PROTECTED] wrote:
Hi, I was wondering if anyone knows of a opensource
SIP
voice logger.
I need to simultaneously record around 150 to 200
sessions.
I figured that if I just set a mirroring port on the
switch and just send all RTP
Can anyone offer any insights as to why with one of these examples I
can do a dial to the sip hone, and with the other I can't?
DOESN'T WORK:
-- Executing Dial(SIP/109-d35d, SIP/101|5|tr) in new stack
-- Called 101
-- SIP/101-c9ff is ringing
-- Nobody picked up in 5000 ms
--
Matt wrote:
In both cases, SIP/116 is on hook and available for calls.
The only thing different is in example one... before it rings
extention 116, it rings extention 101 for 5 seconds.
I know this sounds silly, but you didn't miss anything in the log
stating that the handset had become
Can someone help me with this AGI script to send an email. It just isn't
working. The file is being called in the dialplan and is saved as em.agi
but it isn't sending the email.
#!/usr/bin/php4 -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog =
Nope.. actually, I almost wish the phones did that! (AASTRA 9133i)
because if you are dialing a number, and a call comes in it dumps the
number you were dialing (assuming you didn't finish dialing yet).
It turns out it seems I had a syntax error in my set caller id!!!
Check it out.. notice where
Yes you have a parse error in your PHP when I saved it locally and run it from
the command line I got
syntax error, unexpected '[', expecting ']' in test.php on line 33
Jon FarmerTelford, Shropshire, UK
- Original Message
From: Matthew Warren [EMAIL PROTECTED]
To:
Hi, does anyone know the prices for g729 codecs from Digium? I sent an
email a while ago to them but haven't got any response so far.
Prices are per unit or volume?
Thanks,
--
---
Erick Perez
Linux User 376588
http://counter.li.org/ (Get counted!!!)
Ok that was my fault. I read the diff in the wrong
way!! I found the "ourcallstate" and thought that was the right way. But I have
to patch in the "peercallstate".
Thanks a lot.
Ebse
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Hi,
You can purchase the G.729 codec here:
http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=G729CODEC
The price is $10 / codec
You'll receive the installation procedure by mail,
Regards,
Frédéric Marti
Telecom Engineer
10$/channel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 02, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Prices of g729 codec
Hi, does anyone know the prices for
You can also build G729 codec by urself via Intel IPP.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erick Perez
Sent: Friday, June 02, 2006 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Prices
You can also build G729 codec by urself via Intel IPP.
Regards
===
Do you know if they are compatible with Digium's codecs?
Like this exemple:
2 Asterisk linked via IAX2 , 1 with Intel's codec and 1 with Digium's codec.
Regards
Fred
Hi,
I have an asterisk machine for which the calls reach it via IAX2.
It appears to have a stuck call on it -- there are no channels open, but
one call is active:
linux77*CLI show channels
Channel Location State Application(Data)
0 active
Now why would you want to go and not support Digium and the community for their
hard work to produce a quality product? $10 isnt that much for using
the licenses.. If you take into consideration of how much it COULD cost to
purchase something like this based on circuits it would be insane.
hello, anyone that know about this asterisk's
message:
frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we
already have a VAD frame at the end
Best REgards
Ever
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hello, anyone that know about this asterisk's
message:
frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we
already have a VAD frame at the end
Best REgards
Ever
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hello, anyone that know about this asterisk's
message:
frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we
already have a VAD frame at the end
Best REgards
Ever
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Brian C. Fertig wrote:
Don't cheat digium out of money.. pay the $10 per license.
Yes, be a good colonist and don't dump any more tea into the harbor.
Lee.
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To
Now why would you want to go and not support Digium and the community for their
hard work to produce a quality product? $10 isnt that much for using
the licenses.. If you take into consideration of how much it COULD cost to
purchase something like this based on circuits it would be insane.
We will be closed in observance of the Shavu'ot holiday on Friday, June 2. We
will respond to any messages on Monday morning. Thank you for contacting
CitiPrice!
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On Friday 02 June 2006 11:39, Lee Howard wrote:
Don't cheat digium out of money.. pay the $10 per license.
Yes, be a good colonist and don't dump any more tea into the harbor.
Oh please. Brian's got the reasoning for paying for the license entirely
wrong but at least his heart's in the right
eh? I try..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, June 02, 2006 12:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Prices of g729 codec
On Friday 02 June 2006 11:39, Lee Howard wrote:
On Jun 2, 2006, at 2:59 AM, Thomas Kenyon wrote:
Mike Hammett wrote:
I'm looking for an ATA\Voice Gateway that runs IAX and has several
ports (8 would be nice). I am looking to avoid devices that use the
same firmware as the ATCOM devices as I found them to be buggy (and a
PITA to find the
Hello list,
I have coded something that I call the Asterisk Queue/CDR Log Analyzer.
It is a set of PHP scripts to view, list and graph the QUEUE and CDR log
records. A python script is also included to load the queue log records as
they occur to a MySQL database table. (Asterisk has addons for
On Friday 02 June 2006 11:39, Lee Howard wrote:
Don't cheat digium out of money.. pay the $10 per license.
Yes, be a good colonist and don't dump any more tea into the harbor.
Oh please. Brian's got the reasoning for paying for the license
entirely
wrong but at least his
Does anyone know of a good VOIP provider that I can obtain a DID from Latvia? I live in the US but have a friend there.
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To UNSUBSCRIBE or update options visit:
Interesting. Like I said I've been using them since January and the
only outages we've had have been ones that were planned
maintenance.
George
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
GravesSent: Friday, June 02, 2006 8:36 AMTo: Asterisk
Users Mailing List
Hi Yall,
I would love to put a very compact phone on my wife's desk at work...
Ideally this would be a very small IAX phone with 2 RJ-45's so I could
drop it in without much notice and only have to beg for 1 port from the
sysadmin.
I have looked around and I don't see such an item in
I don't see a problem with monetary reward for hard work. If it wasn't
for Mark, the Digium Team, and the community of developers you wouldn't
have what you have. I am thankful for open source projects and support
in anyway I can.. Money or otherwise. So say I'm brainwashed or
employed either
On Friday 02 June 2006 13:00, Lee Howard wrote:
The GPL very clearly defines the method of expressing deserved
gratitude, and it is not in monetary support.
What's this got to do with the GPL? the g729 codec code is *not* GPLd, and in
fact can't be due to the patent on the implementation of
According to the release notes for Polycom's SIP 1.6.6 firmware the Buddy Watch limitations have been increased from 8 watched buddies to 48 which would give you enough to watch status on three (14 button) side cars.
Haven't tested it but read a discussion in the forum about it and plan to test
try voxbone.com
On 6/2/06, David K Parker [EMAIL PROTECTED] wrote:
Does anyone know of a good VOIP provider that I can obtain a DID from
Latvia? I live in the US but have a friend there.
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I want to overflow calls when all of my local agents are busy and send
up to ten calls to a remote site over TDM (Basically a trunk to trunk
transfer). The catch is that after ten calls I want additional calls to
start stacking up in my local queue and when one of the ten remote
agents
Andrew Kohlsmith wrote:
On Friday 02 June 2006 13:00, Lee Howard wrote:
The GPL very clearly defines the method of expressing deserved
gratitude, and it is not in monetary support.
What's this got to do with the GPL? the g729 codec code is *not* GPLd, and in
fact can't be due to
Oh
sweet.
-Original Message-From: Rob McKrill
[mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 11:25
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Polycom-Asterisk
hints/presence
According to the release notes for
Has anyone done any integration with Asterisk Microsoft Dynamics CRM? I
just wanted to check with the list before I pursue a project with the above
integration. In addition, if anyone would be interested in such an
integration let me know, and I will keep you posted on the results.
Thanks,
I had the same problem with my IAX terminaton provider, when tried to
use codecs other than ulaw (gsm and some others).
Changing back to ulaw fixed the problem. Hope this would help.
Andrei (MPI)
[EMAIL PROTECTED] wrote:
Hi,
I have an asterisk machine for which the calls reach it via
Hello,I am attempting to figure out how to set up SIP trunking, between my company and our SIP provider. This is an expermintal project at this time. The SIP provider gave us a Signalling IP address and two Media IP addresses. We supplied them with the IP address of our Asterisk box. When
If i get a 8XX number, my provider told me that they will send all the
calls he gets. But due to bandwidth and asterisk capacitiy in this
particular installation, the system is only able to handle 27 incoming
calls.
How in my dialplan do I regulate, sending a busy signal, when my
system hits 27
I think you could set the call-limit=27 in either sip.conf or iax.conf, depending on what sort of account setup you have with your provider.i'm assuming that you are going through an ITSP.
On 6/2/06, Erick Perez [EMAIL PROTECTED] wrote:
If i get a 8XX number, my provider told me that they will
Hello,
Thanks again for all the help, and perhaps I have to excuse myself for
replying only after so much time.
I made some progress: some changes in extensions.conf, and removing the
line "register = username:[EMAIL PROTECTED]" makes the
registration timeout errors disappear. I can make
On Fri, 2006-06-02 at 14:56 -0500, Erick Perez wrote:
If i get a 8XX number, my provider told me that they will send all the
calls he gets. But due to bandwidth and asterisk capacitiy in this
particular installation, the system is only able to handle 27 incoming
calls.
How in my dialplan do
Has
anyone got any neat solutions for Asterisk .conf file revision
control?
We
have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common
set of conf files on. They aren't all the same though. There's subtle
differences. For example,in sip.conf, iax.conf etc, the
2006/6/2, Leon Sun [EMAIL PROTECTED]:
10$/channel
If you are connecting a device that uses g729 with another that don't
support it... let's say it uses gsm. Then you will use 2 channels, one
for encoding and one for decoding. Is it?
--
Alejandro Vargas
On Fri, 2006-06-02 at 22:49 +0200, Alejandro Vargas wrote:
2006/6/2, Leon Sun [EMAIL PROTECTED]:
10$/channel
If you are connecting a device that uses g729 with another that don't
support it... let's say it uses gsm. Then you will use 2 channels, one
for encoding and one for decoding. Is
2006/6/2, Lee Howard [EMAIL PROTECTED]:
The messed-up reasoning was that Digium deserves monetary gratitude
because they GPLd Asterisk... and you even said that the reasoning was
wrong.
Digium deserves all money they can win. They are doing a great job,
and they are releasing their job under
I setup a subversion server and a trunk for my different server configs. You might look at that, it does not appear to keep file level versions, but it works great here.On 6/2/06,
Douglas Garstang [EMAIL PROTECTED] wrote:
Has
anyone got any neat solutions for Asterisk .conf file revision
Title: Message
The
first situation you mention can be solved by creating separate files that
contain the unique elements, and then including them in the main files where all
the commonality is. That is how we do things, and it works well for
us. It may be a little cumbersome if you have a
Bruce Reeves wrote:
I setup a subversion server and a trunk for my different server
configs. You might look at that, it does not appear to keep file level
versions, but it works great here.
On 6/2/06, *Douglas Garstang* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Has anyone got
But
you still have to maintain a completely separate copy for each server by doing
that don't you?
That's
what I am hoping to avoid.
It
doesn't keep file level versions? Subversion doesn't do
that?
-Original Message-From: Bruce Reeves
[mailto:[EMAIL PROTECTED]Sent: Friday,
Title: Message
Brad,
Not
sure if #include statments will help. For that to work, there would have to be a
separate directory structure for each server. I'd like to keep it as common as
possible.
If we
had, on our first pbx server...
[general]context=frompstn_startallowguest=yes
Bruce,
Do you
run a subversion client on every Asterisk box, and get the files directly, or do
run the subversion clienton a single central server, and distrubute them
from there?
Doug.
-Original Message-From: Bruce Reeves
[mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006
Alejandro Vargas wrote:
2006/6/2, Lee Howard [EMAIL PROTECTED]:
The messed-up reasoning was that Digium deserves monetary gratitude
because they GPLd Asterisk... and you even said that the reasoning was
wrong.
Digium deserves all money they can win.
I agree entirely.
Note, however,
Title: Message
Ok,
does anyone know if anyone has already created a guide for using subversion with
Asterisk?
I've
hit a wall already, where the subversion docs say that your files _must_ go into
a directory called trunk(huh? What's with that?). That's going to break
Asterisk, who
No, if you do an svn co http://svn.server.com/svn/configs/trunk asterisk
in /etc, it'll make a folder called asterisk in your /etc directory. Once
that's done, any modifications made that are committed to the server can
be downloaded into /etc/asterisk by running svn up inside the directory.
I use subversion on a central server and then store each server that is different. The purpose behind it for me was 2 fold, first I have a backup of my configs centeralized and I can roll-back any changes. Second, I can checkout a servers files on a different machine to edit them if I want and
On Fri, 2006-06-02 at 14:18 -0700, Lee Howard wrote:
Andrew took issue with my initial sarcastic comment because this thread
involves the G.729 codec - but remember that if someone does ultimately
choose to obtain a license illegally that they're not cheating *Digium*
- rather, they're
Aaron,
I'm trying to check-in (is that the right term?) the files for the first time.
There's nothing in the repository yet.
Doug.
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Friday, June 02, 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial
Is there a way to limit the size of a Queue? I want to create a queue
with for example, 5 agents, and only allow at most 10 persons waiting
so this way, they don't saturate my entire PSTN span, which can be
also simultaneously used for another Queues or for my outgoing calls.
Thanks a lot for
Bruce,
But,
if you have three servers that function the same, don't you have to check the
file out three times and check it back in three times?
Doug.
-Original Message-From: Bruce Reeves
[mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006
3:34 PMTo: Asterisk Users Mailing
Wow, I went through and setup an acoount but couldn't figure out why I could only get a fast busy signal when dialing, then, checking the FAQs I noticed this very important detail, you cannot place outgoing calls over Voxbone. How messed up is that? They don't bother to mention this fact in there
On Saturday 03 June 2006 09:37, Douglas Garstang wrote:
Aaron,
I'm trying to check-in (is that the right term?) the files for the first
time. There's nothing in the repository yet.
http://svnbook.red-bean.com
hads.
___
--Bandwidth and Colocation
Are you following the quickstart in the SVN book? For the first time to import them in to a folder called trunk. Then as Aaron stated you can check or co the trunk to any folder.
On 6/2/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Aaron,I'm trying to check-in (is that the right term?) the files
Read this:
http://subversion.tigris.org/faq.html#repository
http://svn.collab.net/repos/svn/trunk/README
That'll link you to the README that comes with subversion, which has a
very detailed explanation on how to get a repo set up and running :) If
it says anything in there about using trunk,
If all 3 servers are the same then no. I import to the svn server the check out the files on each server. I f I change a file on server A I can then commit the change to the repository, on the central server, and then do a svn update on the other 2.
On 6/2/06, Douglas Garstang [EMAIL PROTECTED]
-Original Message-
From: Hadley Rich [mailto:[EMAIL PROTECTED]
Sent: Friday, June 02, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Config Revision Control
On Saturday 03 June 2006 09:37, Douglas Garstang wrote:
Aaron,
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