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
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 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 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
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
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 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
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 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 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
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
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 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
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
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 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
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
That debug looks cool but I have no idea what it means.
If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor.
When you can hear the audio of the fax hopefully you will be able to
tell what's going on, and if you're lucky it's something specific to
the particular kind of
read the 1.6 README and the 1.8 README.
If you're using SIP you should expect changes with account
authentication, faxing, output regarding channel status and
performance.
I think that version of 1.4 is late enough you would already be on
DAHDI for hardware devices. If not, you need to convert
I'm having annoying errors trying to get configure working.
tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz
cd asterisk-1.8.6.0
./configure
I get complaints related to pwlib / ptlib...
checking for openr2_chan_new in -lopenr2... no
checking /root/pwlib/include/ptlib.h usability... no
checking
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming kpflem...@digium.com wrote:
This is a bug in the configure script, but in the meantime, you should be
able to use --without-pwlib to avoid it, as long as you aren't trying to
build chan_h323.
Thanks much.
I was trying
./configure
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
However you could select/deselect modules using menuselect if you wanted to
automate the process. It's documented over here:
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439
On Mon, Sep 12, 2011 at 11:19 AM, Tarek Sawah tareksa...@hotmail.com wrote:
i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed
at 100 concurrent calls.
please advise?
Nobody will know why your asterisk crashed unless you follow the
instructions here:
On Wed, Oct 19, 2011 at 7:19 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
Thank you, I actually found the asterisk.conf settings after sending the
mail. So next question is which folders/files do I need to change ownership
of to make it work?
/etc/asterisk
On Sun, Oct 23, 2011 at 3:16 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote:
If you use DAHDI, you need to change ownership of /dev/dahdi/* to the
non-root owner. I ended up rolling that into the init script for
dahdi
On Thu, Oct 27, 2011 at 11:53 AM, Mike l...@net-wall.com wrote:
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
On Thu, Nov 10, 2011 at 12:24 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-11-10 12:12 PM, Danny Nicholas wrote:
Yeah! My boss will be much happier having a system that doesn't have the
-tail on it.
I hear this kind of statement every once in a while, which makes absolutely no
On Fri, Nov 18, 2011 at 2:23 PM, Sazzad sazzadbinka...@gmail.com wrote:
Hi,
I have to use asterisk with some dedicated DSP chips, which will do the
expensive G729 CODEC computing, so that the server processor has minimum
load. I was informed, I've to use GPAK to implement this. So far I've
I
On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas da...@debsinc.com wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
What's the difference between a car released in 2006 versus a car
On Thu, Jan 5, 2012 at 8:05 AM, Steve Underwood ste...@coppice.org wrote:
No PAP2 or PAP2T supports T.38, even though many people will swear that they
do. For a little while there was some beta code for the PAP2T with badly
broken T.38 support. Perhaps this is where the legend of T.38 on a
On Wed, Jan 4, 2012 at 4:45 PM, Asterisk Development Team
asteriskt...@digium.com wrote:
The Asterisk Development Team is pleased to announce the first
release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.
2.6.0 is a feature release which:
wct4xxp: Expose serial number in dahdi_device and
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie
m...@chrishowie.com wrote:
I've been up and down this issue for a few hours and I cannot for the
life of me determine why simply defining a peer causes Asterisk to offer
telephone-event. I have tried specifying dtmfmode=rfc2833 or
On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib fkha...@iconnecths.com wrote:
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is
called but it doesnt execute the command, I tried the command in terminal it
worked, any help please ... below is my dial
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg dbackeb...@gmail.com wrote:
shebang /path/to/bash
PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug
On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards
asterisk@sedwards.com wrote:
The OP was using MIXMONITOR_EXEC (although I wonder about the '' syntax)
so he doesn't need to explicitly execute (via system()) his commands.
Wow. Never knew that was possible. I still don't like the syntax, but
On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote:
Hi everyone,
Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about
5000 numbers and then put the call to agents right away and pull up the CRM
based on the number dialed. So, I am going to be
401 - 440 of 440 matches
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