On Tue, Apr 13, 2010 at 7:43 PM, Prince Singh pri...@drishti-soft.comwrote:
1. Are Asterisk and Mittel in the same physical LAN.. or do they have a
router between them?
2. Do a 'rtp debug' at the Asterisk CLI to see where is the RTP data
being sent to
3. Probable issues:-
Reading of all the brute force attacks on the list i was wondering if
anyone has implemented some connection rate limiting rules in Shorewall to
stop the brute force attacks?
I'm a bit puzzled about which rule(s) to use on which ports, if
anyone could help me with some example rules to start
Per Jessen wrote:
Just start it with safe_asterisk.
http://linux.die.net/man/8/safe_asterisk
Unless my info is out of date, it will kill two birds with one stone.
Asterisk will restart itself, and you will get a core dump.
Thanks,
Steve Totaro
Hi Steve
I've got three such core
On Tue, Apr 13, 2010 at 10:43:41PM +0100, Gordon Henderson wrote:
On Tue, 13 Apr 2010, Asterisk Development Team wrote:
* Static /dev/dahdi files are not generated at install time since udev is
used
on all the supported distributions. build_tools/make_static_devs is
available for
On Tue, Apr 13, 2010 at 05:46:03PM +0100, Ricardo Coelho wrote:
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2
asterisk nodes sharing every configuration file (by using a distributed
filesystem) except the master.csv files. If asterisk does not keep
On Tue, Apr 13, 2010 at 04:25:49PM -0600, John Rose wrote:
Why do versions 1.6.2 and 1.6.1 use much more CPU resources that 1.6.0?
I can get 400+ SIP/G.711
calls running on this dual core box with 1.6.0 but the cpu maxes out and
core dumps at approx. 180 calls when version 1.6.1/2 is
Hello Huu,
use E1/SS7 signaling or if you MSC speak SIP, then use SIP.
Thanks
On Tue, Apr 13, 2010 at 11:46 AM, huu giang huugiang...@yahoo.com wrote:
Hi all,
My Asterisk connect to GSM core network (connect directly to MSC) through
E1 lines. What the kind of protocol is used ?. It is
Hi Goke,
Some experienced people said me to use ISDN to connect to MSC.
Thanks very much.
--- On Wed, 4/14/10, Goke M Aruna gok...@gmail.com wrote:
From: Goke M Aruna gok...@gmail.com
Subject: Re: [asterisk-users] protocol used to connect Asterisk and GSM core
network (MSC)
To: Asterisk
hello Huu,
Can you share their explanation with me at least, I can gain from it too.
Thanks
On Wed, Apr 14, 2010 at 10:01 AM, huu giang huugiang...@yahoo.com wrote:
Hi Goke,
Some experienced people said me to use ISDN to connect to MSC.
Thanks very much.
--- On *Wed, 4/14/10, Goke M
I'm running 1.6.1.18 on an older ubuntu machine. I upgraded to
dahi-linux-2.3.0. That went fine, and it works. But I decided to use the
opportunity to upgrade to 1.6.2.6. That didn't work.
configure, make menuselect, make, make install all went fine, or at
least seemed to. But it hangs
You can take a look here http://www.gl.com/images/GL_Network_GSM.gif .
As far as I know Asterisk (FreeSwitch, Yate or what so ever for PBX
system) is normally connected to the GSM core network thru the GMSC
(Gateway MSC). They talk with each other SS7 ISUP.
Theoretically seen one might have a
I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me. I would like to know
what
This sound strange, i have running on asterisk 1.4
1090 calls with no problem 24/24 h.
(24 giga ram 4 dual core xeon )
Maybe is the configuration or configuration tuning missing in somewhere.
Andrea
Il 14/04/2010 10:45, Tzafrir Cohen ha scritto:
On Tue, Apr 13, 2010 at 04:25:49PM -0600, John
http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, April 14, 2010 10:52 AM
To: asterisk-users@lists.digium.com
Subject:
Luki,
Thanks for the quick response -- that did exactly what I was looking for and
was even easier than I thought.
Lincoln
--
Lincoln King-Cliby, CTS
Applications Engineer | Crestron Certified Programmer (Silver)
ControlWorks Consulting, LLC
V: 440.449.1100 x1107 | F: 440.449.1106 |
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might be a better way of doing it.
On Wed, Apr 14, 2010 at 11:08 AM,
Tonty-
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might be a better way of doing it.
Can you explain the multiple
How many simultaneous conference meetme setups can be supported in the same
time on Asterisk, and what are the corresponding server's specs for this.
Thanks--
_
-- Bandwidth and Colocation Provided by
Hi!
I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I
need a lot of script done in perl and I wanna move to windows server. I checked
Asterisk::fastagi but I see that everything is for Linux.
Somebody has idea to do this in perl. I dont want to change the language.
Tonty-
This is more or less the idea. I was not thinking about the E3 then break
it down, because I am not sure they provide E3s, they suggest me invest into
multiple E1 cards to support as many call as I can
Ok but how do you get the data? 35 E1s is a lot of cabling for an external
On Wed, 14 Apr 2010, torinti...@hotmail.com wrote:
How many simultaneous conference meetme setups can be supported in the
same time on Asterisk, and what are the corresponding server's specs for
this.
How long is a piece of string?
0) A better subject yields better answers
1) A more
On Wed, 14 Apr 2010, Edwin Quijada wrote:
I wanna know if can run my AGI scripts as fastAGI scripts in Windoes
server.
Seems like a move in the wrong direction to me, but no
-- you can't run an AGI script via fastagi() without changes.
I need a lot of script done in perl and I wanna move
Hi there,
Any of you know how can i configure two PBX servers with FreePBX with load
balancing? Do I need a SIP server like OpenSER? Any of you have some
configuration reference?
I have installed AsteriskNow in both FreePBX servers.
Thank you!
Regards
--
Dear Sir,
I'm trying to configure the SMS capabilities of my provider (Telecom
Italia) with my asterisk. I tryed with a normal phone (SMSC number
42100) and all is working fine, incoming and outgoung SMS.
Now I'm trying to configure my asterisk for support SMS, the incoming
messages are working
Hello,
I can reproduce the results always.
Hardware:
E-5200 2.5 Ghz
4 gig ram
Centos 5.4
I've tested most versions from 1.6.0.11 to 1.6.0.26.
Compared to most version of the 1.6.1.x and 1.6.2.x.
I originate calls from one box using FastAGI to another Asterisk box
that is the call sink.
On Wed, Apr 14, 2010 at 10:33 AM, Tonty T ton...@gmail.com wrote:
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might
Last year I did a lab test for a customer who wanted conferencing solution
for his organization, on a 2 x dual core xeon with 4GB type server, which
had 120 zap channels and I put all the channels in mutiple conferences, from
4 to 20 users per conference and let it running for two weeks. Munin
On Wed, Apr 14, 2010 at 10:33 AM, Tonty T ton...@gmail.com wrote:
This is a solution they proposed, using GSM gateways, but it wont let me
handle 1000 simultaneous calls, the other option was using an E1 but the
cost would be too much to deploy 35 E1s to support that many calls. There
might
I need the server to handle about 300 - 400 simultaneous meetme conferences,
5-10 participants in each,
Actually I need to know, if I will get an IBM X3650 M2, QuadCore, 4-6 GB RAM,
8MB cache,
how many simultaneous meetme conferences I can operate on a this server.
Thanks
From: Zeeshan
Hi listers,
I'm a 1.4 holdout who uses mostly Suse platforms. I bit the
bullet and installed a Centos box to get Lumenvox up and running, but now
see this new offering of Vestec that supports OpenSuse and Windows in
addition to the Platforms supported by Lumenvox. Anybody out
That's is all the overhead I am trying to avoid. What I need is a DID with
unlimited channel, but they do not offer DIDs in that country. I wanted to
know for example when I get a DID from lets say Vitelity, with unlimited
channel, what are they using to forward the calls via SIP or IAX to my
Hi Guys,
i've connecting two pbx server successfully for several times using the
following config:
register = USPBX:myp...@122.11.176.35 uspbx%3amyp...@122.11.176.35
[PBX1]
type=friend
host=122.11.176.35
trunk=yes
sercret=mypass
context=external
deny=0.0.0.0/0.0.0.0
Hi guys,
I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.
So each ppl will have a audio and video stream.
I'm wondering if you know a way to do this with asterisk or if it's
supported
Hi!
I need the server to handle about 300 - 400 simultaneous meetme
conferences, 5-10 participants in each,
Actually I need to know, if I will get an IBM X3650 M2,QuadCore, 4-6
GB RAM, 8MB cache, how many simultaneous meetme conferences I can
operate on a this server.
There is no
Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
Hi guys,
I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.
So each ppl
On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland baula...@epitech.net wrote:
I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.
Else, do you know any other way to do this ?
Hi all,
What would be the best way to send a call to a queue as usual, but telling
that it should be awsered by some specific member?
Thanks already
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hello list,
I am new to this list and have been using Asterisk as part of my research
project for about 2 weeks now.
I would like to get your thoughts on a scenario that I am attempting at the
moment. I haven't had luck until now.
In this scenario, I am trying to register a SIP user
My problem is that I need to execute windows app using IVR in Asterisk so we
need FastAGI using perl. I saw Asterisk::fastagi but everything for this is in
Linux and i dont know if it works in windows.
I need to know if somebody has used fastagi in windows with perl becuase I have
a lot
On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada
listas_quij...@hotmail.com wrote:
My problem is that I need to execute windows app using IVR in Asterisk so we
What is the windows app that you cannot replace on Linux?
How about wrapping THAT program with simple inputs and outputs, and
build a
Dear Goke,
I don't use ISDN to connect to MSC, it connect to ISDN network.
There are other people deploy IVR using this protocol.
About ISUP/SS7, supporting technical from the vendor I bought Sangoma Card,
they said that If I want to connect to MSC, I have to use ISUP/SS7 protocl, but
now
Hello,
I wanna record the conversations in a conference call, anyone know how can I do
it? I've already configurated a room on meetme.conf but I don't know as I can
record the conversations.
I'm using SUSE 11 and Asterisk 1.6.2.
Thank you so much for help me.
Bye
Hi Huu,
Asterisk support ss7.
Check chan_ss7 and libss7, both project are active and working like charm.
Thanks
On 4/15/10, huu giang huugiang...@yahoo.com wrote:
Dear Goke,
I don't use ISDN to connect to MSC, it connect to ISDN network.
There are other people deploy IVR using this
http://www.projectdiastar.org/ looks promising...
On Apr 14, 2010, at 7:04 PM, Stéphane Bauland wrote:
Le 04/15/2010 12:11 AM, Hans Witvliet a écrit :
On Wed, 2010-04-14 at 22:55 +0200, Stéphane Bauland wrote:
Hi guys,
I'm planning of creating a speech/video conference application. This
We have used with success BBB (BigBlueButton - open source -
http://bigbluebutton.org) and I recommend to try their demo in order
to see if this solution gives all you need.
Voice conf is based on Asterisk.
HTH,
Ioan Indreias
www.modulo.ro
On Thu, Apr 15, 2010 at 2:04 AM, Stéphane Bauland
On 4/15/10 1:26 AM, Tonty T wrote:
That's is all the overhead I am trying to avoid. What I need is a DID
with unlimited channel, but they do not offer DIDs in that country. I
wanted to know for example when I get a DID from lets say Vitelity,
with unlimited channel, what are they using to
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