Russel,
Please excuse me for saying it yet once more... (look for the thread Stable
Stable Asterisk, from Sunday). Build bots are nice to check and spot for
compile errors (which is good). But I think that what people are looking here
(well, specially me) is a set of automated tests for all of
Ditto. Would you complain if some one gave you a free flight that it wasn't
first class ? Asterisk is free Stop the moaning
Enough The Digium/Aseterisk bashing seems to be at an all time high
recently. You seem to be involved in a lot of it. Russell has given most of
Users register to (Open)SER which uses same DB as all Asterisk nodes.
Asterisk Realtime engine lets change data in only one database to make
changes global. (Open)SER does load-balancing and fail-over.
You can even put second (Open)SER server in case first dies and use DNS SRV
to make it active.
You can try MOR FREE billing system for Asterisk.
LiveCD can be downloaded from:
http://www.kolmisoft.com/mor/index.php?option=com_contenttask=viewid=73
Regards/Pagarbiai,
VoIP Billing Solutions
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED]
You can try this: http://www.voip-info.org/wiki-NVFaxDetect
Regards/Pagarbiai,
Mindaugas Kezys
VoIP Billing Solutions
http://www.kolmisoft.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, August 29, 2007 4:32 PM
To: Asterisk Users Mailing List
snip
Digium has done this, for me, as well.
However, in either case, I have reservations about letting others wack
away at my machines, especially if one cannot see what they are doing.
No so much not trusting them, but not learning a thing along the way.
When I voiced that concern to the Digium
Hi Russell,
First of all, let me tell that in my company only buy Digium Cards, because:
- Is the company founded by Mark Spencer, and buying Digium hardware is
a way to support Asterisk (in my opinion)
- Since today I only can tell good things about Digium: good support to
the comunity, good
From our testing, the TE120P will fix the issue.
Best of luck,
PaulH
On Thu, 2007-08-30 at 09:19 +0200, Marc Patino Gómez wrote:
Hi Russell,
First of all, let me tell that in my company only buy Digium Cards, because:
- Is the company founded by Mark Spencer, and buying Digium hardware
Dear Kezys;
Thanks a lot, but from where I can have the
configuration manual (to know how to configure it).
Regards
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
You can try MOR FREE billing system for Asterisk.
LiveCD can be downloaded from:
Hi All,
Twice now in the past few weeks I've walked into the office to find that
our 1.2.24 Asterisk process is sat at 100%, and that hundreds of
thousands of log files in /var/log/asterisk exist, all at 312 bytes,
containing:
Aug 29 23:22:17 VERBOSE[24303] logger.c: Asterisk Event Logger
Oh, you have *got* to be waiting for someone to make a joke about
asterisk crashes vs. plane crashes. No?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: 30 August 2007 08:07
To: Asterisk Users Mailing List -
On Thu, Aug 30, 2007 at 10:15:43AM +0100, Adrian Marsh wrote:
Hi All,
Twice now in the past few weeks I've walked into the office to find that
our 1.2.24 Asterisk process is sat at 100%, and that hundreds of
thousands of log files in /var/log/asterisk exist,
I suspect misconfigured
I am trying to retrieve the dialed peer number but
it seems that ${DIALEDPEERNUMBER} is broken.
Also, I know that I could extract the dialed number
from the ${CHANNEL} variable but this only works for
SIP and maybe IAX (untested). However, it doesn't work
for ZAP. All I get when using ZAP is
In article [EMAIL PROTECTED],
Adrian Marsh [EMAIL PROTECTED] wrote:
Hi All,
Twice now in the past few weeks I've walked into the office to find that
our 1.2.24 Asterisk process is sat at 100%, and that hundreds of
thousands of log files in /var/log/asterisk exist, all at 312 bytes,
Absolutely not! However, if someone gave me a free flight, but the
plane went down 3 out of the 5 times it took off, yes I would :)
Then, if the makes of the plane released a new version where they
fixed the problem, but now instead of going down because the motors
shut off, it would go down 3
Thanks for the answers Tony/Tzafrir
I checked the disk usage stats, and they are constant throughout the
period. I have a script that runs through on-the-hour to clean out
recordings 3hours old, and I monitor disk usage via SNMP.
I wonder though if a log file could also cause this? Maybe the
MAC = Move Add Change..
On 8/29/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Awesome, when you say end user do you mean the people sitting at the
phones or the person doing MACs, or both?
people at the phones: yes
the person doing MACs: what is that? (your
Hi,
How can I have A*k convert a call from +441793xx to Dial
00441793xx instead?
With the _+. Below I can catch the call, but EXTEN doesn't get set
as expected.. and then I need to figure out how to pass the call onto
the outgoing-pstn context. Not sure if a Goto would work here...
On 8/29/07, BJ Weschke [EMAIL PROTECTED] wrote:
I think we will want to see what state chan_sip is sending into
app_queue for it to be called Uknown. What is the last state these
channels are in before they go to Unknown in app_queue?
Unfortunately, I don't know. This is in an active call
On 8/30/07, Adrian Marsh [EMAIL PROTECTED] wrote:
[outgoing-pstn-international]
exten = _+.,1,Set(EXTEN=00${EXTEN:+1})
exten = _+.,2,NoOp(test line: ${EXTEN})
Setting ${EXTEN} won't work, but Goto(context,00${EXTEN:1},priority) will:
[foo]
exten = 7997,1,Answer
exten =
On Thu, Aug 30, 2007 at 12:17:49PM +0100, Adrian Marsh wrote:
Thanks for the answers Tony/Tzafrir
I checked the disk usage stats, and they are constant throughout the
period. I have a script that runs through on-the-hour to clean out
recordings 3hours old, and I monitor disk usage via SNMP.
Hola lista.
Tengo un pequeño problemilla. explico. :
Tengo dos asterisk, conectados entre si. Un Ast1 hago el resgistro de todos
mis clientes y en el otro Ast2 termino las llamadas por Zap. Todas las
llamadas que se realicen de Ast1 externas ( _X. ) van a Ast2 y si llaman a un
numero que
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
I am trying to retrieve the dialed peer number but
it seems that ${DIALEDPEERNUMBER} is broken.
Also, I know that I could extract the dialed number
from the ${CHANNEL} variable but this only works for
SIP and maybe IAX (untested). However, it doesn't
ok tnx guys
On 8/25/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Le Fri, 24 Aug 2007 20:50:05 +0400, Mark Quitoriano
[EMAIL PROTECTED] a écrit:
What is a good softswitch that is also open source rather than asterisk?
You may want to check out freeswitch.
Many of these issues only appear when you put it into production and/or
after a period of time. Most of the crashes I've seen are like this. I
simply to not have the resources to run simulations to try to find these
types of issues.
I can do one of several things. I can simply not upgrade
On Wed, Aug 29, 2007 at 04:37:14PM -0500, Russell Bryant wrote:
Steve Totaro wrote:
I don't see Matt as a troll, he is mostly helpful to people on these
lists (if memory servers me correctly).
Kind of harsh for am employee of Digium on a public Asterisk mailing
list, don't you think?
Russell Bryant wrote:
Steve Totaro wrote:
I don't see Matt as a troll, he is mostly helpful to people on these
lists (if memory servers me correctly).
Kind of harsh for am employee of Digium on a public Asterisk mailing
list, don't you think?
I tend to make my passes through the
Hello,
Looks like I have been able to get the jain-sip-phone to work. The problem
seemed to have been an sdpFactory.createconnection call. It was passing one
parameter, which was the IP Address. I had to change this to the call with
three parameters (ie: sdpFactory.createconnection(IN,
On Thu, Aug 30, 2007 at 07:15:51AM -0400, Matt wrote:
Absolutely not! However, if someone gave me a free flight, but the
plane went down 3 out of the 5 times it took off, yes I would :)
Then, if the makes of the plane released a new version where they
fixed the problem, but now instead of
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess that's my point. I realize asterisk is open source and FREE,
however, I wouldn't expect a commercial application to crash as often
as I've seen asterisk go down.
Windows 98.
Cheers,
-- jra
--
Jay R. Ashworth
Matt wrote:
I guess my request is just that Digium maybe spend a little more time
in QA before rolling a release out the door. It's just annoying when
you do what should be a dot upgrade, and find out a feature that had
worked just one dot below has now stopped working, or worse yet
On Wed, Aug 29, 2007 at 10:33:31PM -0500, Russell Bryant wrote:
Brian West wrote:
I commend these efforts but if it compiles it doesn't mean it won't
crash in certain conditions much less run at all. Proper unit testing
is hard to do trust me I have been reading up on the subject and in
Matt wrote:
Absolutely not! However, if someone gave me a free flight, but the
plane went down 3 out of the 5 times it took off, yes I would :)
Then, if the makes of the plane released a new version where they
fixed the problem, but now instead of going down because the motors
shut off, it
On Thu, Aug 30, 2007 at 09:38:46AM +0300, Diego Iastrubni wrote:
I have a been working on such a list, but it's more or less concentrated on
channel banks (like duh... look at my email...). I would be more then happy
to give you the list of tests I have made if you desire.
I did start a
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jay R. Ashworth
Sent: 30 August 2007 13:57
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] where is 1.4.12?
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess
In article [EMAIL PROTECTED],
Adrian Marsh [EMAIL PROTECTED] wrote:
Thanks for the answers Tony/Tzafrir
I checked the disk usage stats, and they are constant throughout the
period. I have a script that runs through on-the-hour to clean out
recordings 3hours old, and I monitor disk usage via
So don't take the free ticket. Go with another solution. There are hundreds
of thousands like myself that are happy with asterisk. No one is forcing you
to do it. (If it is the boss's then it's your job and their head ache. If
they complain about the reliability explain them that you get what
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
- Original Message -
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 3:57 PM
Subject: Re: [asterisk-users] where is 1.4.12?
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess that's my point. I realize asterisk
I agree with Russell's initial assessment; Matt's phrasing, if not his
intent, emanated from the land of the troll. . . if for no other reason
than the implication that Digium is solely responsible for the
development of the product.
I want to reply to this my initial comments were not
Agreed.. and as I stated in the post I just made people WILL go to
other solutions if they get a poor taste of Digium/Asterisk it is
in Digium's best interest to try to work as many bugs out of the free
version as possible.
On 8/30/07, Dovid B [EMAIL PROTECTED] wrote:
So don't take the
On Wed, Aug 29, 2007 at 09:48:18PM -0400, Matt wrote:
I guess that's my point. I realize asterisk is open source
and FREE,
however, I wouldn't expect a commercial application to
crash as often
as I've seen asterisk go down.
Windows 98.
wouldn't expect != haven't
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
I have a SIP phone calling via a SIP trunk another asterisk system, that then
sends the call out a ZAP channel.
When I press any of the features defined in features.conf, The end user on the
ZAP side hears the DTMF tones, and none of the features work.
My DTMFmode on the SIP users definition
On Thursday 30 August 2007 9:49:57 am Matt wrote:
I want to reply to this my initial comments were not trolls.
I think, however, my initial comments reflect what alot of the
asterisk community is experiencing.WE support asterisk for people.
WE also sell phone systems based somewhat
On Thu, 2007-08-30 at 08:02 -0500, Eric ManxPower Wieling wrote:
As I understand it, Digium does NO formal QA testing before the free
Asterisk/Zaptel/libPRI releases. Asterisk Business Edition is a
different story and gets extensive QA testing.
As I understand it, that's simply due to a
Thanks James, worked a treat.
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
[local]
exten = _${SOMEVAR}.,1,NoOp(Dialled own number)
I'm looking to catch our own PSTN numbers outbound should someone use the full
PSTN number rather than the local
Either way, setting up mysql-ndb is not hard.
-bk
- Original Message -
From: Andrew Latham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 6:22:37 AM (GMT-0600) America/Chicago
Subject: Re:
Dear All,
How long should it take before a exten = h,1,Hangup() kicks in,
versus a exten = s,n,Hangup()
I'm just about to test, but thought I'd ask.
--
http://www.suretecsystems.com/services/openldap/
___
--Bandwidth and Colocation Provided by
I'll admit I've been bitten once or twice by bugs AFTER a rollout, the vast
majority of my installations work, as far as the customer is concerned.
Yes.. OUR rollouts work fine, because we use a version of asterisk
that we are comfortable with. However, I'm talking about when we do
consulting
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 = _${SOMEVAR}.,1,NoOp(Dialled own number)
No, unfortunately you can't use variables as part of the extension name
or pattern match.
On Aug 30, 2007, at 10:11 AM, Jared Smith wrote:
On Thu, 2007-08-30 at 15:42 +0100, Adrian Marsh wrote:
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
[local]
exten = _${SOMEVAR}.,1,NoOp(Dialled own number)
No, unfortunately you can't use variables
/32-1 is ringing
-- Accepting call from '2177' to '7141278' on channel 0/30, span 1
-- Executing Monitor(Zap/30-1,
gsm|/asterisk/out-velton-20070830-181638-7141278-2177-1188486998|bm)
in new stack
-- Executing Set(Zap/30-1, CALLERID(all)=1759) in new stack
-- Executing Dial(Zap/30-1
Yes.. OUR rollouts work fine, because we use a version of asterisk
that we are comfortable with. However, I'm talking about when we do
consulting for someone who has installed their own asterisk and then
they have some issues with it...
This is the problem to use the last release of
Brandon Kruse wrote:
Either way, setting up mysql-ndb is not hard.
You're correct, it's a matter of hours.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -
Guillermo,
Me parece que la cosa aqui es que el nombre del usuario debe ser el mismo
en el URI del fuente que en el el proceso de autentificacion.
Traiga poner username= en la configuracion asi:
On Thu, 30 Aug 2007, Guillermo Rodriguez wrote:
[pbx1]
name=test1
callerid=200
host=dynamic
Tried the AsteriskNOW beta6 VMWare image yesterday. It's come a long way
since last time I looked at it a few months ago. Some things are nicely
polished and worked very smoothly but somethings were surprisingly flaky.
Maybe because I never had all the incoming/outgoing/user stuff configured as
Just IMHO but you shouldn't be doing regular updates on a phone system that
is working well unless you are doing it to fix a specific problem. It's a
phone system not a server. I mean security upgrades as well. At least not
until they have been out there for a considerable amount of time. Yea
Then you should probably use a commercial application like the Business
Edition. I've found that once I decide to go down the open source road it's
a different ball game. Test with the latest and greatest release that has
the features you need. If it's a fairly new release chances are it's not
To match any single digit use X. Also, it is simplest to know what your
+ meta is for and just match that. In the states we just match _011X.
Anthony
Adrian Marsh wrote:
Thanks James, worked a treat.
Is there a way of using variables within the dialplan, eg:
[globals]
SOMEVAR=0179344
Agreed, unless the security vulnerability could allow calls to be made
to premium rate service numbers that charge $500/min. Obviously, you
could have the telco block international (speaking as a person inside
the US) dialing.
Also, you have the disgruntled employee, ex-employee, or customer
Hi:
I purchased TDM40B from a week ago from digium, iam
trying to install it to the current version of zaptel
(zaptel-1.4.5), but when i make modprobe wctdm these
came out :
[EMAIL PROTECTED] root]# modprobe wctdm
FATAL: Module wctdm not found.
FATAL: Error running install command for wctdm
Jared Smith wrote:
On Thu, 2007-08-30 at 08:02 -0500, Eric ManxPower Wieling wrote:
As I understand it, Digium does NO formal QA testing before the free
Asterisk/Zaptel/libPRI releases. Asterisk Business Edition is a
different story and gets extensive QA testing.
As I understand it,
shadowym wrote:
Then you should probably use a commercial application like the Business
Edition. I've found that once I decide to go down the open source road it's
a different ball game. Test with the latest and greatest release that has
the features you need. If it's a fairly new release
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Russell Bryant [EMAIL PROTECTED] wrote:
If the TE110P will not work out for you, Digium will trade it for a TE120P.
The
120 is the replacement for the 110 which uses a far superior PCI interface
developed at Digium instead of the
The Digium cards are known to steal IRQ's.
The Sangoma cards do not.
Arthur Miller
Sr. Sales Associate
VoIP Supply, LLC.
454 Sonwil Drive
Buffalo, NY 14225
716-250-3871 OFFICE
716-630-1548 FAX
[EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED]
NOTICE: The information
Stephen Bosch wrote:
Jared Smith wrote:
On Thu, 2007-08-30 at 08:02 -0500, Eric ManxPower Wieling wrote:
As I understand it, Digium does NO formal QA testing before the free
Asterisk/Zaptel/libPRI releases. Asterisk Business Edition is a
different story and gets extensive QA testing.
As I
On Thu, Aug 30, 2007 at 09:59:22AM -0700, jonny hashem wrote:
Hi:
I purchased TDM40B from a week ago from digium, iam
trying to install it to the current version of zaptel
(zaptel-1.4.5), but when i make modprobe wctdm these
came out :
[EMAIL PROTECTED] root]# modprobe wctdm
FATAL: Module
--- Atis [EMAIL PROTECTED] wrote:
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
I am trying to retrieve the dialed peer number
but
it seems that ${DIALEDPEERNUMBER} is broken.
Also, I know that I could extract the dialed
number
from the ${CHANNEL} variable but this only works
for
SIP
Arthur Miller wrote:
The Digium cards are known to steal IRQ's.
The Sangoma cards do not
Not to appear defensive, but that is a technically inaccurate and also
technically ambiguous statement. To correct it, there used to be a
potential problem related to using the TE2xxP/TE4xxP cards
Hi:
I'm doing my first PRI order for a client in Western Canada, and I have
the initial setup questionnaire in front of me. It has about 25
questions on it.
Some of it I understand, most of it I don't. If there are any Canadian
list members out there who have ordered PRI recently and who are
This is actually a big misconception... the idea that you don't need to
match + because you'll never receive a + and it's just a metacharacter.
In the modern world of IP phones and such, more often than not, you will
ACTUALLY be sent a + and will need to translate that yourself on your
own
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
However, I'm still having some trouble trying to
understand why Asterisk does not log the ZAP number as
is otherwise the case for SIP. A description of this
problem is at:
http://lists.digium.com/pipermail/asterisk-users/2007-August/193645.html
I
What phones are you using?
SIP wrote:
This is actually a big misconception... the idea that you don't need to
match + because you'll never receive a + and it's just a metacharacter.
In the modern world of IP phones and such, more often than not, you will
ACTUALLY be sent a + and will need
Hello all,
please, can anyone advertise me some channel banks, which can
operate with E1 (30 FXS)? Rack-mountable option is welcome. I've
tried to google, but I've not found nothing appropriate. More
than one E1 link is welcome too. Everyone channel banks, which
I've found, was between T1 and
Snom, UTStarCom, and the usual assortment of softphones (X-Lite,
SJPhone, Snom360 Softphone, eyeBeam, Bria).
N.
Anthony Francis wrote:
What phones are you using?
SIP wrote:
This is actually a big misconception... the idea that you don't need to
match + because you'll never receive a
hi:
Iam using Mandrake 10.1 ,with kernel 2.6.8.1-12mdk
and here where are zaptel.ko:
/lib/modules/2.6.8.1-12mdk/misc/zaptel.ko
/lib/modules/2.6.8.1-12mdkcustom/extra/zaptel.ko
thanks
jonny
Building a
http://www.wired.com/print/politics/security/news/2007/08/wiretap
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Bruce McAlister wrote:
Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to
initialize G.729 copy protection!
Hi,
Could anyone from Digium please shed some light on the build
environment for the solaris 10 g729 codec?
Was it build on Solaris or OpenSolaris?
Are there any
--- Atis [EMAIL PROTECTED] wrote:
On 8/30/07, Vieri [EMAIL PROTECTED] wrote:
However, I'm still having some trouble trying to
understand why Asterisk does not log the ZAP
number as
is otherwise the case for SIP. A description of
this
problem is at:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
So, now that we've all complained about the state of testing of Open
Source versions of Asterisk, lets do something about it.
I propose we start with a list of things that we think should be tested
in Asterisk, and means to test them.
Maybe we
Anyone remember the problem with writing out config files that had
#include directives in them? You'd get a single, flat config file when
you saved it back out. I have heard a few howls of complaint! (see bug
8684)
Well, I just checked in a fix for this into trunk. My simple tests say
it's
On Thu, 2007-08-30 at 14:13 -0600, Steve Murphy wrote:
2. Blank lines between entries will get dropped. Sorry. If you really
like
blank lines, then include a comment of a blank line.
Ouch... this makes it quite cumbersome. In fact, that's the number one
complaint I get from students in the
Matt Riddell wrote:
Should these tests be added to Asterisk-Addons or maintained outside of
the tree?
If people start writing test utilities, I would be happy to host them in a
subversion repository. Depending on the size of this stuff, it could probably
go into the main Asterisk repository.
For anyone who is interested in the solution:
It seems Asterisk detected a busy signal. Setting 'busydetect=no' in
zapata.conf solved this problem.
Lars
--
Sure there have been injuries and deaths in boxing,
but none of them serious.
-- Boxer Alan Minter
Indeed very interesting and informative. I think this has been covered
in past issues of 2600, but this is the first time these docs are
available.
Thank you
On 8/30/07, Joe Acquisto [EMAIL PROTECTED] wrote:
http://www.wired.com/print/politics/security/news/2007/08/wiretap
Can you explain this question?
Just to clearify, exten = h will execute as soon as Asterisk is aware
that the channel was hung up. While app_hangup will execute a hangup
on an active channel.
On 8/30/07, Gavin Henry [EMAIL PROTECTED] wrote:
Dear All,
How long should it take before a exten =
Hi All -
Has anyone had a chance to use the Asterisk Appliance yet? Any
thoughts or reactions? I have a couple of clients waiting on the
Zaptel version, but maybe somebody has used the VoIP-only version?
Thanks,
Noah
___
--Bandwidth and Colocation
- Original Message -
From: Jared Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 5:41 PM
Subject: Re: [asterisk-users] where is 1.4.12?
On Thu, 2007-08-30 at 08:02 -0500, Eric
Are you using 1.4.X on one and 1.2.X on another ?
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, August 02, 2007 3:34 PM
Subject: [asterisk-users] problem with rfc2833
I have the following:
pri box incoming/outgoing on box
How about sending a SipHeader to the second box and then on the second box look
for the header. If the header does not exist then ring the extension normally.
If the header is there than send back congestion (basically have a gotoif
before it hits the Exten = Foo,1,Voicemail)
- Original
You may want to consider upgrading your version of asterisk. Next you can try
using SER + Asterisk + Heartbeat.
- Original Message -
From: Khaled Chehab
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL PROTECTED]
Sent: Tuesday, August 21, 2007 3:05 PM
snip
question2: it's possible read registration data from astdb from python/php
(or it is possible write sip registrations to mysql/sqlite? i do not
want realtime because of NAT issues)
/snip
Marek,
What NAT issues can realtime create that there won't be in static ?
I am a long time user and reseller of Thirdlane PBX Manager. From my
standpoint the implementation tools are outstanding and the fact that
the files are easy to follow means it allows a consultant to comstomize
the behavior yet allow the end user to maintain going forward.
good luck!
Steve
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 28, 2007 11:51 AM
Subject: Re: [asterisk-users] OT: DELL Platforms
Dovid B wrote:
snip
I am running an SC1435
- Original Message -
From: Eric ManxPower Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Wednesday, August 22, 2007 4:08 PM
Subject: Re: [asterisk-users] Polycom behind NAT won't register to
- Original Message -
From: Adrian Marsh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 30, 2007 2:34 PM
Subject: [asterisk-users] How to handle + prefix
Hi,
How can I have A*k convert a call from
I was wondering if anyone has an easy way to emulate dialing in a round
robin fashion like when you use Zap/r1 for Zap trunks. At the moment
what I do is simply make a macro that will dial the sip trunks in order
so if the first one fails it goes to the second and so on. The problem
with
Bruce McAlister wrote:
Bruce McAlister wrote:
Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to
initialize G.729 copy protection!
Hi,
Could anyone from Digium please shed some light on the build
environment for the solaris 10 g729 codec?
Was it build on Solaris
On Thu, 2007-08-30 at 16:35 -0400, Jared Smith wrote:
On Thu, 2007-08-30 at 14:13 -0600, Steve Murphy wrote:
2. Blank lines between entries will get dropped. Sorry. If you really
like
blank lines, then include a comment of a blank line.
Ouch... this makes it quite cumbersome. In fact,
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