currently chosen on the other
call leg. The problem with this is that we are not guaranteed the call leg
supplying the codec will not change later.
--
David Vossel
Digium, Inc. | Software Developer, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
(0x2b6fc145d000)
Any thoughts on why this is happening as I could not find many
references to it when searching ?
--
Thanks, Phil
Do you have res_jabber installed?
--
David Vossel
Digium, Inc. | Software Developer, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
asterisk@sedwards.com wrote:
Many times, I've made the statement that you can execute hundreds of AGIs
written in C in the time it takes to load an interpreter and parse a script
written in PHP or Perl.
I've truly enjoyed this thread. And
hI
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http://www.asterisk.org/hello
asterisk-users mailing list
To
On Mon, Jun 27, 2011 at 9:06 AM, Michael voip.quest...@gmail.com wrote:
Hi Kevin,
Controlling it through the sip.conf peers is sufficient for us for this case
(because this particular provider doesn't support T.38 at all), but I think
it would be a good idea to add the option to
Hi all,
We have a project involving voice recognition, and will need a vocabulary of
10,000 words (actually names).
Can anyone recommend a product that works with Asterisk?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
-pickup-db70;2'
status is 'UNKNOWN'
The context doing the pickup looks like:
[product-pickup]
exten = _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone)
Thanks for any advice,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0
in debugging this issue ?
David
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk
Just as a note, the versions of ConfBridge in 1.8 and Trunk are completely
different. Trunk will likely give you much better results. In 1.8 ConfBridge
is more of just an experimental exercise of the bridging API.
--
David Vossel
Digium, Inc. | Software Developer, Open Source Software
445 Jan
Jared,
Thank you for that information!
Has anyone else had an experience like this?
On 12 May 2011 20:25, Jared Geiger compuw...@gmail.com wrote:
Hi David,
When I was testing 1.6.1 for high volume channels, I couldn't get over 1000
channels / 40 CPS without the load average spiking up
Hello,
We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
experiencing higher CPU utilization on their server. I can't see anything
wrong, so is this just expected with 1.6? Can anyone help explain it?
Thanks for any advice.
--
David Cunningham, Voisonics
http
On Thu, May 5, 2011 at 1:43 PM, vip killa vipki...@gmail.com wrote:
The majority of open source projects out are NOT run by commercial
institutions...
Postfix kicks butt. But only because IBM paid for development, for a
long number of years, and because they hired somebody who had a really
good
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a
and will give you lots of distortions on your
VoIP.
David
On 2011-04-28 11:25, Bruce B wrote:
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all
other bad quality things that can happen to a SIP trunk? I have plenty
of bandwidth and crisp clear lines so the only
wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David
*Sent:* Thursday, April 28, 2011 10:32 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re
:-*
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl
exist?)
pri=1
asterisk=
asterisks=127
mysql=
mysqls=127
On Wed, Apr 27, 2011 at 8:43 PM, Juan David Diaz juanch...@gmail.comwrote:
Hi:
http
at a time because I want to validate the user's entry
at each key press.
Thanks,
David
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New to Asterisk? Join us for a live introductory webinar every Thurs
Hi:
http://php.net/manual/en/function.system.php
Then, the commands you shoul run:
/usr/sbin/asterisk -rnxpri show spans
/etc/init.d/asterisk status
/etc/init.d/mysql status
.
.
.
.
and so on!!
good luck!
Regards.
Juan.
Linux User #441131
On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati
for closing hours.
David
On 2011-04-27 13:34, vip killa wrote:
I just completed building a feature rich asterisk voicemail system
using perl, php, and mysql.
My only concern is that the system i built will not be able to handle
the call volume needed. Let me start by explaining my setup
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe with modern DAHDI.
Will I be able to use this on any virtual server without having the need
special changes to
the VM
- Original Message -
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2011 9:27:19 AM
Subject: Re: [asterisk-users] The new ConfBridge application is now in
Asterisk Trunk
- Original Message -
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 25, 2011 9:49:05 AM
Subject: Re: [asterisk-users] The new ConfBridge application is now in
Asterisk
? or only 1.10?
Jerry
It will be introduced in Asterisk 1.10.
Please don't create a new thread for questions regarding the new ConfBridge
application when there is one already going on.
~David
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-- Bandwidth and Colocation
signalling between the two calls.
Maybe something is different.
What I find really weird is that the DTMF is incorrectly sent from the
first asterisk only when the second asterisk bridges to DAHDI.
Any ideas?
David
On 11-04-23 11:48 AM, David wrote:
Hello,
I installed Asterisk 1.6.2.17.3
version.
Everything else is identical. So the problem appears to be caused in the
RTP and not in the SIP. So something about the RTP packets coming from
the DAHDI channel on asterisk-pri makes asterisk server send invalid DTMF.
David
On 11-04-24 11:42 AM, David wrote:
I did more testing.
Here
__ast_read: DTMF end
emulation of '#' queued on SIP/omnity-0023
I notice that the # key was repeated several times by the DTMF even
though the dialplan only calls # once. Why are these two different when
the DTMF sequence is exactly the same ?
Any ideas?
David
and do not know where to go from here. I would really
appreciate it if someone could give me some pointers on where to go next, what
additionnal debugging steps I should perform. I would also really appreciate if
someone could propose a solution.
Please help!
David
Never give up, never surrender
ON
* *** test program only **
*
* (c) David Cook, 1994
*
* Set signlal lines on serial port to turn on 5vdc
* signal. Used for solid-state relay (low current
* draw on RS232C port) to switch high voltage/high
* current load
http://www.google.com/search?q=port+5000+asterisk
answer is in the first hit :)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Thomas
Sent: Wednesday, April 13, 2011 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial
On Fri, Apr 1, 2011 at 7:04 AM, Khaled W. Chehab kche...@xplorium.com wrote:
1-Is there a way to export fax tiff file image from .pcap captured file .
Maybe, but I can't think of how. If you can somehow invert the pcap
file back into packets and reproduce the fax traffic, then maybe.
In other
on version 1.8 that I haven't seen yet?
Any comments or ideas are greatly appreciated!
Thank you!!
--
David Cabrejos
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Only by replacing it.should not be a problem.
Juan.
Linux User #441131
On Wed, Mar 9, 2011 at 8:13 AM, Satish Patel satish...@hotmail.com wrote:
Hey guys,
Currently we have non HWEC sangoma pri card but now we are planing to
replace card with HWEC support card for echo cancellation. So
On Mon, 7 Mar 2011 08:50:27 -1000, Matt Darnell
mattdarn...@gmail.com wrote:
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote:
Dear
this note is only for fresh administrators don't think about asterisk
security.
Do you know where you go to 'un-ban' an IP if they made some
Dean,
what's your zaptel Zapata config_
regards
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote:
We are running Asterisk version 1.4.23-1, libpri-1.4.9 and
zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10.
We have a T1 PRI
group=33
context=testivr-in
signalling=fxo_ks
threewaycalling=yes
transfer=yes
channel=96
---
On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com
wrote:
Dean,
what's your zaptel Zapata config_
regards
Juan.
Linux User #441131
Hoover kb7...@gmail.com wrote:
This doesn't represent the 2nd span?
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
Dean
On Mon, Feb 21, 2011 at 1:18 PM, Juan David Diaz juanch...@gmail.com
wrote:
I don't see any problem.. but, i don't see the 2nd SPAN @ zaptel:
yellow alarm on span 2
Ooops, my bad I Did not read the zaptel config file correctly, my apologize.
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
*span=2,1,0,esf,b8zs
bchan=25-47
dchan=48*
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:34 PM, Juan David Diaz juanch...@gmail.comwrote:
Your message
have you check the PRI crossover cable?
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:35 PM, Juan David Diaz juanch...@gmail.comwrote:
Ooops, my bad I Did not read the zaptel config file correctly,
my apologize.
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
*span=2,1,0,esf,b8zs
bchan=25
system.conf:
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3
Hi,
Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have
terrible sound: the MOH is unrecognizable and speakers can't be
understood; it sounds ghostly. However the prompts (your are the only
one in this conference, etc.) sound fine.
Our server has a Digium T410P card with two E1
On Tue, Feb 08, 2011 at 10:59:47AM -0600, Danny Nicholas wrote:
Any idea?
I use mpg123 to play my MOH so I can control the volume (my users complain
that standard MOH is a bit loud).
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with
On Tue, Feb 08, 2011 at 11:09:19AM -0600, Warren Selby wrote:
On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org wrote:
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with MusicOnHold(default). So it's
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.)
mcolu...@sirioinformatica.it:
Hi, I have asterisk 1.6.2.6 on a Debian Lenny system.
When I try to send a fax in T.38 mode I receive this error
ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel
'SIP/eutelia-sirio-out-' is in an
Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI or a SIP trunk with the following configuration
Dell R710
Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single
Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz
/drivers/dahdi/dahdi-base.o] Erreur
1
make[1]: *** [_module_/usr/src/dahdi-linux-2.4.0/drivers/dahdi] Erreur 2
make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.34.6 »
make: *** [modules] Erreur 2
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0
Shaun,
CONFIG_MODULES wasn't enabled - thanks for the advice!
On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote:
On 1/30/11 8:45 PM, David Cunningham wrote:
I'm installing Asterisk with Dahdi on a server with a custom kernel
compile. I've got the kernel source
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman brya...@zktech.com wrote:
Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't
make t.38 work I keep getting the following error Disconnected after
permitted retries Any ideas on this?
So you're saying if you turn off
On Tue, Jan 25, 2011 at 9:34 AM, Bryant Zimmerman brya...@zktech.com wrote:
On 01/24/2011 2:54PM Bryant Zimmerman wrote
The attached file was too large so I am putting in a link to the file. It is
a virus free text file.
You failed to mention earlier that this is T.38.
Turn off T.38 and see
On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman brya...@zktech.com wrote:
Do you know how to force off T.38 in res_fax?
it's in sip.conf
take a look for
t38pt_udptl=yes
change it to no
reload sip
on your console
that should force it to either fail entirely or do audio passthrough.
--
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote:
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries.
How would
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
We know the problem exists -- the boss just installed U-verse at his house
:)
It works fine from cell and copper, just not from U-verse and their ilk.
Well, I would say more data samples are needed then. It could
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal ami...@phoenixinternet.net wrote:
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can
send recieve faxes from both boxes fine to and from pstn. But the faxing
between 1.6 and 1.4 extensions does fail. Any ideas please ?
You
.
If you have any questions please don't hesitate to contact me directly.
Regards,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
% increase) that would be great, rather than just
lots.
Also, are there any ATAs which are known to not work with progressinband =
yes? We have Polycom, Linksys and Audiocode.
Thanks for any advice,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20
Each media stream will use two, one for RTP and one for RTCP. In your case
10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is
always and even numbered port, and RTCP is always RTP port + 1. Yes, it's
in the RFC for RTP.
The fact that you have two pairs means that two
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham my...@techsol.org wrote:
For some reason our Asterisk box is doing something really unusual following
applying a routine update to CentOS 5 on Monday.
We have Asterisk 1.4.2 and its been working great for years. But now when
the phone system
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com wrote:
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to
posted also this on the following forum:
http://forums.digium.com/viewtopic.php?f=1t=76529sid=dc731d8832e381826ba8c5da6483bdd1
--
David Cabrejos
--
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New
On Wed, Dec 8, 2010 at 9:06 AM, Gilles codecompl...@free.fr wrote:
Hello
I need to find a recent and neutral comparison of the major products
available to connect an Asterisk server to the telephone network,
whether ISDN (BRI) or PSTN, and through a PCI card or some external
box. I'm
On Wed, Dec 8, 2010 at 10:17 AM, Gilles codecompl...@free.fr wrote:
On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg
dbackeb...@gmail.com wrote:
* pay somebody else to do it in the form of appliance and lose most
control versus do it yourself and have total control but also the
chance to screw
AGI program.
Thanks for any advice!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
--
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-- Bandwidth and Colocation Provided by http
Steve, that looks just the job, thank you very much.
On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 7 Dec 2010, David Cunningham wrote:
Is it possible to somehow 'bookmark' a place in a sound file? That is,
the user presses a key while a sound file
On Tue, Nov 23, 2010 at 8:25 AM, voip crazy voipcr...@gmail.com wrote:
Hello,
I want to analyze the asterisk logs files, looking for all kind of
errors, ¿Anyboby knows any asterisk logs analyzer?
You're only going to have the logs for what you create logs for.
I create custom logs for the
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson duane.lar...@gmail.com wrote:
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
If you remove an extension that
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call
time...but I simply don´t know what could it
be.
Thanks for your help!!
Kind Regards.
Juan.
Linux User #441131
On Wed, Nov 17, 2010 at 3:59 AM, Alexis de BRUYN ale...@de-bruyn.fr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Juan David,
While starting heartbeat service
on by first time...but I simply don´t know what could it
be.
Thanks for your help!!
Kind Regards.
Juan.
Linux User #441131
On Wed, Nov 17, 2010 at 3:59 AM, Alexis de BRUYN ale...@de-bruyn.frwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello Juan David,
While starting heartbeat
Hi Asterisk Users,
I´m in trouble setting a HA cluster for Asterisk service.
While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel
asterisk* on haresources), zaptel and asterisk does not start as
I´m expecting, this is the debug result:
ResourceManager[3260]:
Juan.
Linux User #441131
-- Forwarded message --
From: Juan David Diaz juanch...@gmail.com
Date: Tue, Nov 16, 2010 at 1:38 PM
Subject: HA - asterisk service is not starting
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi Asterisk
On Mon, Nov 15, 2010 at 8:30 AM, Richard Kenner ken...@gnat.com wrote:
It's kind of low for me. How does one control that volume?
I've never heard of a way to control that volume.
You can tweak after-the-fact with sox, or you can crank up your
soundcard / amplification on playback.
--
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch ca...@usawide.net wrote:
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far away, and process the calls?
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into
for any help.
David
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing
On 10/21/10 12:07 PM, Steve Howes steve-li...@geekinter.net wrote:
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
I'll subscribe, that is for sure. What is the best way to dist the
blacklist? iptables include file? Or something more integrated to
asterisk... just thinking off the top
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question voip.quest...@gmail.com wrote:
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote:
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
You don't say anything about what you're trying to send / receive against.
Here's how you should troubleshoot:
* start with a 'real fax machine'
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst fivehor...@gmail.com wrote:
How did the setup work as far as extensions on the Inter-Tel system
contacting extensions on the asterisk system?
It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel
worked. Still had to watch out for
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question voip.quest...@gmail.com wrote:
The whole point (as I specified in the header and initial message) is the
attempt to use Fax for Asterisk to send the message.
Asterisk can handle audio passthrough faxing. I'm talking audio faxing
over SIP. You
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question voip.quest...@gmail.com wrote:
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll dan...@danielknoll.de wrote:
Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this for the current channel.
Only the command meetme list roomnr shows the usernumber, but i can't
On Wed, Oct 6, 2010 at 5:00 PM, marvin horst fivehor...@gmail.com wrote:
Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
system via a SIP trunk using the IPRC card?
I have, believe it or not, integrated Asterisk with Inter-Tel.
However, not via SIP. Run the costs.
On Thu, Sep 30, 2010 at 10:51 AM, khalid touati khalidtou...@gmail.com wrote:
Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i came accross this isssue: i can't load the fax module:
pbx3*CLI module load res_fax_digium.so
Unable to load module
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati khalidtou...@gmail.com wrote:
thanks for replies,
I am using Asterisk 1.6.2.11
and components res_fax-1.4_1.2.1-x86_64 and
res_fax_digium-1.4_1.2.1-barcelona_64.
(amd 64 bit machine)
actually I am not aware that there is version which include
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
The reason is when doing a load balancing , We cannot confine the
recording to a particular asterisk machine ( If we have more than one
asterisk machine in the topology ).
Yes you can. You can
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote:
Don sez: I don't know how to make Outlook indent. I usually top-post, but I
don't like getting yelled at.
Why do you say Don't do that? Is there a real reason that it would be bad?
Performance is a real reason. Multiple
Danny, thank you!
On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham
*Sent:* Wednesday, September 22, 2010 4:28 PM
*To:* Asterisk Users
On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
HI ,
Is there Any way is there so that I can store my recordings directly to a
database rather storing the same to a file .
Please, please, please tell us why you would want to do that.
--
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett a...@plexicomm.net wrote:
In the simplest terms I can think of, I'm going to describe what I want to
do and I want to know if it's possible in the current version of asterisk.
Can I take a T38 call from an ATA, convert that back to analog and have
All,
Two questions:
1. Is there a limit on how long a call can be recorded for? For example is 4
hours a problem?
2. Can recording be stopped after a configured period of silence?
Thanks in advance,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Now I have no previous experience with Cisco systems and don't want to screw
up anything. Are they much different than Asterisk based systems? I guess
the underlying VoIP technology is the same for both the systems so it
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Now I have no previous experience with Cisco systems and don't want to screw
up anything. Are they much different than Asterisk based systems?
sometimes. Cisco supports SIP, but depending on the product,
asterisk
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote:
Can you recommend any specific solution to this problem or way to install
app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
You don't say the percentage that are failing. However, people
I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf. I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf.
One channel is routed to the new context as I want, but the other
channel is stuck
On Tue, Aug 24, 2010 at 9:05 AM, Ron nha...@gmail.com wrote:
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i used it as a sip trunk for my asterisk. i'm a newbie when it comes to
ISDN. and i've been experiencing some issues:
1. Call Hangup:
When hangup is
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull dun...@e-simple.co.nz wrote:
Voice recognition is a pain for people with accents and poor lines and when
Everybody has an accent. Some people live in a place where the people
they talk to sound like themselves, so they forget that fact.
Of
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote:
We gave the phone a static IP address and pointed it to the configuration
server on the remote end that has the CFG files for it. The phone starts
up, downloads SIP and the “new application” and otherwise seems to be
On Fri, Aug 13, 2010 at 11:43 AM, Eric Merkel (Mail Lists)
ejmerkel.li...@gmail.com wrote:
I am looking to build a small PBX for an office that has 3 incoming analog
lines and less than 10 extensions.
For that small of an installation you might prefer an asterisk
appliance. You can review the
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