2009/2/27 Alistair Cunningham acunning...@integrics.com
Ignacio,
Our Enswitch product matches all these requirements; indeed it goes well
beyond them:
- We scale far beyond 3000-4000 concurrent calls. We'd consider such a
system medium sized. At this size the system is fully
hi guys:
recently i shoule use the meetme2 application and use the webto control ,but i
can find the way to install it. I have research a lot of articls but have
nothing improved.
could some one give me a help article or sime helpful web uri? I will
appreciated!
my asterisk is 1.4.2 ,thanks
2009/3/1 michel freiha mich...@gmail.com
Dear David,
I'm using G729 pass though mode...No transcoding is used here
Regarding concurrent calls, I have 3 asterisk servers working in load
balancing mode...The issue that the same problem appear on 3 asterisk...each
asterisk handle around 150
You could try our qloaderd - it was made for MySQL, but it should be simple
enough that by changing the engine should be a no-brainer (it's a perl
script). You get the added advantage that in case anything goes wrong with
the DB system, you lose no data.
It is here:
Hi,
I am testing some IP phones (eg. GXP2000) and noticed that the early dial
feature works fine with Asterisk 1.4 but not with 1.2.
early dial is when digits are sent immediately, one by one, and Asterisk
replies with a 484 Address Incomplete and waits for the next digit until a
match is
Hi all,
I've been using telephone cards for 4-5 years and now I'm considering
the Patton gateways. What are the pro and cons of Patton stuff compared
to internal cards in a production system? Someone say external gateway
are better because has no echo but have a longer delay when placing
Hi,
The part of pedantic=yes that you need to make '#' work is URL
encoding, unfortunately it comes with a whole load of other baggage
that breaks a lot of different things. A simple fix might be to
comment out the parts of pedantic=yes that you do not need in the
source code and re-compile -
Hello all,
Not sure if this mail belongs to this users or dev list. Sorry about
that.
We have the following scenario:
PhoneA OpenSER AsteriskPhoneB PhoneC
| | | | |
| |
2009/3/2 Giorgio Incantalupo gincantal...@fgasoftware.com
Hi all,
I've been using telephone cards for 4-5 years and now I'm considering
the Patton gateways. What are the pro and cons of Patton stuff compared
to internal cards in a production system? Someone say external gateway
are better
Only need to make change in file.c :
int ast_language_is_prefix = 0
And that's all!
2009/2/27 Giedrius Augys voi...@gmail.com
Hi,
I have problem with Asterisk 1.6.0.1. I need to change language for
playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime
plays in English,
On Monday 02 March 2009 00:27:00 Steven J. Douglas wrote:
Hi,
My asterisk segfaults a few times each day and the crash problem seems
weird. When I run gdb on the core dump, it almost always segfaults on
free() or malloc(). When I run the back trace, I see something weird.
Here's one of the
Hi Olivier,
so if you say that Patton hardware is bad documented, hard to configured
and without echo canceller I think it is useless...don't you think?
Unless it is much more reliable (no crashes at all)
Olivier wrote:
2009/3/2 Giorgio Incantalupo gincantal...@fgasoftware.com
Hey, all. I was going through a make configure on my Asterisk 1.4.23
Ubuntu box, and noticed something I'd forgotten: Asterisk now supports
IMAP_STORAGE. However, when I highlight it, it tells me that there's an
unmet dependency, presumably for imap_tk. I've apt-get installed
everything I can
On another topic, I would say those gateway are not so easy to configure :
- a web server is embeded but it is not documented anywhere and it's
GUI is far from natural,
- alternatively, you can edit a config file for which a huge doc is
available but, as this boxes are not specifically
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
Thanks,
-Ken
--
This message has been scanned for viruses and
2009/3/2 Giorgio Incantalupo gincantal...@fgasoftware.com
Hi Olivier,
so if you say that Patton hardware is bad documented,
hard to configured
and without echo canceller I think it is useless...don't you think?
I won't say so.
Would you say Bristuff is well documented ?
What I meant is I
2009/3/2 Darrick Hartman dhart...@djhsolutions.com
On another topic, I would say those gateway are not so easy to configure
:
- a web server is embeded but it is not documented anywhere and it's
GUI is far from natural,
- alternatively, you can edit a config file for which a huge doc is
Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
asterisk -help
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
On Mon, 2 Mar 2009, Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
man asterisk, read about
Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
Thanks,
-Ken
Run Asterisk with the -g
You could change your asterisk command to asterisk -vvg, but this will eat
your disk space if you have a large number of faults since each core.* file
produced takes up 1-13 Mb.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ken D'Ambrosio wrote:
So: what/how do I need to install to meet this dependency?
Did you run configure again after installing the missing components?
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
Ken D'Ambrosio wrote:
So: what/how do I need to install to meet this dependency?
You need to read the documentation, specifically doc/imapstorage.txt,
which is conveniently located in the source tree and named with a name
very similar to the feature you are trying to use :-)
--
Kevin P.
Thanks! I'll give that a try.
Regards,
Steve.
Tilghman Lesher wrote:
On Monday 02 March 2009 00:27:00 Steven J. Douglas wrote:
Hi,
My asterisk segfaults a few times each day and the crash problem seems
weird. When I run gdb on the core dump, it almost always segfaults on
free() or
Hi Ken,
If you run ulimit -c on the command line and get a 0 output, then
you need to run ulimit -c unlimited on the command line.
-Steve
Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to
Thanks Steve.
I guess I could set pedantic=no but modify chan_sip.c, add a global variable
such as urlencode that if set to yes will call:
ast_uri_encode
ast_uri_decode
and nothing else.
I wish I didn't have to hack the source code though. I'd rather make the sip
client work somehow.
Danny Nicholas wrote:
You could change your asterisk command to asterisk -vvg, but this will eat
your disk space if you have a large number of faults since each core.* file
produced takes up 1-13 Mb.
In the day and age where 500GB hard drives are $75 at Micro Center, hard
drive space
You would think this, but I've seen asterisk create 100 or more dumps in an
hour of 10+Mb. Depending on Inode size, etc., this situation could push a
system into a hurting capacity rather quickly. Also, many shops use older
technology and compound this by RAID striping, which can reduce your
Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to debug? I looked
in the obvious places in make menuconfig and didn't see anything
appropriate.
There should be a file called something like
What's a reasonable bounty to put on this?
I would like the functionality that was in 1.4.19-1.4.21(?) returned.
I was holding out on the bug, waiting for a resolution, but it looks
like the community would rather redo the way CDR works although, then
patch a broken, flawed system. In the
Hi All,
Have any way to disable Asterisk to not generate a ring tone when
receive a 180 message and use only the early media in 183 ? My service
provider send 180 first and after 183 message and asterisk start
generate ring when receive 180 and when receive 183 get the ring from
SDP this
Hello, Im using channel_h323 by Jeremy McNamara to connect my asterisk box
to an Gatekeeper and I want to do some filter by remote ip addres but I
dont know what variable in asterisk have this data. Someone knows how is
the name or which are the name of this variable in channel h323? Thanks for
Danny Nicholas wrote:
You would think this, but I've seen asterisk create 100 or more dumps in an
hour of 10+Mb. Depending on Inode size, etc., this situation could push a
system into a hurting capacity rather quickly. Also, many shops use older
technology and compound this by RAID striping,
I have a PRI line and I am having problems setting the ringtimeout on the
dial application to more than 29.
If I set ringtimeout to 29 on the dial application call and I do not answer
the ringing phone then I correctly get DIALSTATUS set to NOANSWER.
If I set ringtimeout to any value over 29 on
Hi all,
I'm using asterisk in real time mode...All extensions are defined in table
sip_buddies...Everything looks fine and asterisk is reading extensions info
from the sip_buddies table...The problem occurs as soon as any information
on an extension is changed from sip_buddies table...Which mean,
Jim Dickenson wrote:
If I set ringtimeout to 29 on the dial application call and I do not answer
the ringing phone then I correctly get DIALSTATUS set to NOANSWER.
What kind of phone?
If I set ringtimeout to any value over 29 on the dial application call and I
do not answer the ringing
On Mon, 2009-03-02 at 22:31 +0200, michel freiha wrote:
Hi all,
I'm using asterisk in real time mode...All extensions are defined in
table sip_buddies...Everything looks fine and asterisk is reading
extensions info from the sip_buddies table...The problem occurs as
soon as any information
Dear Sir,
The issue has been solved
rtcachefriends=no
\and everything will work
Thanks
On Mon, Mar 2, 2009 at 10:31 PM, michel freiha mich...@gmail.com wrote:
Hi all,
I'm using asterisk in real time mode...All extensions are defined in table
sip_buddies...Everything looks fine and
On Fri, Feb 27, 2009 at 7:49 PM, Jared Smith jsm...@digium.com wrote:
On Fri, 2009-02-27 at 14:07 -0700, Brandon B. wrote:
As of yet, I am unwilling to change the LBO to 0 to where it probably
should be because the system is working and I'm not sure exactly what
the LBO does. I'm aware
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Kevin P. Fleming kpflem...@digium.com
Organization: Digium, Inc.
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 02 Mar 2009 14:41:51 -0600
To:
On Sat, 28 Feb 2009 21:52:46 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sat, Feb 28, 2009 at 11:24:53AM -0800, Michael Higgins wrote:
So, tweaking configs, rebuilding this and that... restarting,
twiddling, it works (yeah!), but fails on re-boot to work at all.
Consistently,
How many call do you think it will handle?
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Hi all,
I'm experimenting with the MySQL realtime logging of queue log, using
backports for 1.4. When I configure the logging of queue log as
realtime, the previous way to save it onto the plain text file stops
working. Is there a way to make it work both ways, just like the CDR
works, where
Hi all,
I'm using asterisk in real time mode...My extensions.conf table contains:
[default]
switch = Realtime/defa...@extensions
I have added the following to extensions.conf table;
context:micho
exten: _X.
priority: 1
app:Dial
appdata: SIP/00xxx...@pstn GAteway
Asterisk server is connected
Any way to initiate a call and execute a playback of an audio file from the cli?
My only chance to debug or make changes is usually when no one's at the office
including me!
Thanks!
jlc
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Have a look at 'call files' on voip-info.org
Great fun, especially for load testing.
PaulH
Joseph L. Casale wrote:
Any way to initiate a call and execute a playback of an audio file from the
cli?
My only chance to debug or make changes is usually when no one's at the
office including
I would like to do the following:
Dial an extension in Asterisk
The extension runs an application which dials a number (like a hybrid of
DIAL and READ). The dialed number is a box which does nothing but play
DTMF tones then hangs up
The first box captures the DTMF tones to a variable
Dial
Hi,
Is there a way to do a blind transfer within an asterisk dialplan (like '##')?
The reason I need this (I think) rather than a regular Goto() is that
I'm trying to do one-touch parking.
I can park a call using one-touch parking and then pick it up again,
however if I try to re-park the call, it
I tried to get some of that stuff going a while ago, and it just didn't
work - like the polycom user status stuff (at lunch) Asterisk sees the
bits of info but doesn't want to handle it.
PaulH
lord_f...@iinet.net.au wrote:
hi all,
i have 2 x-lite version 3.0 softphones configured on
first thing to do is
switch =Realtime/@
then closely check which extension ur dialing.
means length of extension u define in appdata 00XX
its length is 8 and if u dialed and extension greater then 8 or less then 8
then u cant go through...
for simplcity u check it with
appdata:
Have a look at 'call files' on voip-info.org
That worked well.
Thanks!
jlc
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On Tue Mar 3 11:38 , Paul Hales sent:
I tried to get some of that stuff going a while ago, and it just didn't
work - like the polycom user status stuff (at lunch) Asterisk sees the
bits of info but doesn't want to handle it.
PaulH
hi Paul,
Thanks for the reply, its good to know someone else
Not many.
Ignacio Ortega A. wrote:
How many call do you think it will handle?
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