Displaying 20 results from an estimated 9000 matches similar to: "Limiting Simultaneous calls"
2007 Sep 18
4
Linux limits
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for "asterisk1/700"
Too many open files
Is this a limit of my Linux box? I only have 512MB of ram. Will increase
it to 2G help or I have to change some configuration in
2008 Sep 27
3
test call generator
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each route. As well as do test
at every interval too
If you know something like this please enlighten me.
Sam
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2008 Aug 20
1
vicidial mysql problem
I installed asterisk, astguiclient, php and mysql. but when i dialled one
number to another number my asterisk server give the following error:
> /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
> install_driver(mysql) failed: Can't load
>
'/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so'
> for module DBD::mysql: libmysqlclient.so.15:
2008 Oct 05
5
asterisk, phpagi and singleton
Hello,
I've this situation: 300+ simultaneous calls and dialplan like this:
exten => _X.,1,Answer()
exten => _X.,2,DEADAGI(check_status.php)
exten => _X.,3,Dial(SIP/other/${NUMBER})
exten => _X.,4,Hangup
exten => h,1,DEADAGI(cdr.php)
When project is running , I had a lot of defunct php scripts (I've exceed
mysql connection limits and so on, deadagi help a bit). The
2008 Sep 30
3
Maybe OT - routing calls in PSTN
I have a Vitelity DID which generally works, but calls from a particular
caller do not reach it. Vitelity has thus far disavowed any
responsibility for working through this problem. I recognize that some
action might be required by another provider which is outside Vitelity's
control, but it seems that they should at least be trying to help
resolve the problem by helping me determine
2007 Sep 25
2
Point-to-Point SIP link without registration
Greetings list,
I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls.
One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either
2008 Jul 19
2
OT Astricon/Digium Beach Ball Mailing
Just an FYI for Digium. I received a mailing today from you guys
which was nice. The price of mailing was ~$1.60 and inside was an
inflatable beach ball.
Cool, but I tried to blow up the beach ball and the the seam where the
part opens to inflate the ball was not connected to the ball
whatsoever, so it went right in the trash.
I wonder if the sick heat had anything to do with it, was mine just
2008 Sep 15
6
Callcenter monitoring tool
Hello all,
Anyone expecialized with call center monitoring and reporting solution
based on asterisk.
A client of us, want to install a call center reporting solution for
an asterisk server but I do not know which could be the best tool for
that.
I need a tool for reporting queue calls, agent calls, and disconnect cause.
Any clue will be appreciated.
Thanks in advance.
VoipCrazy
2008 Sep 18
4
OT - How to stream a A-Law/wav file to a browser ?
Hi,
How can I create a web page allowing people to listen (with their own PC) a
couple of .wav/a-law files stored on a Linux server ?
Chances are users would access this web page from Internet Explorer but if I
could make it available to other browsers, that would be better.
I googled a bit and couldn't find a tag such as media://myaudiofile.wav that
would fulfill this spec.
As much as
2007 Sep 14
4
how to route outgoing calls on IP-level
Dear Sirs,
out asterisk server has multiple network cards.
I want some outgoing calls (from several extensions) to use one IP address,
and others to go through
another address.
is there a way to achive that using asterisk ?
Cheers,
Kate
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2009 Oct 18
4
Astricon
Wish I could have made it :( Is there a possibility of a collection of
the talks/slides/handouts/videos/presentations for download? Even pay
for?
Cheers,
j
2008 Oct 27
11
Fring: Open VPN client to be installed on the mobile, which mobile?
Hi All;
I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile:
Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server?
Regards
Bilal
2007 Sep 25
5
Do I need to run #modprobe zaptel for Digium
Hi List;
If I am configuring Diguim Analoge card, then I need
to run #modprobe wctdm, but the question why I need to
run #modprobe zaptel also?
What #modprobe zaptel does a things that #modprobe
wctdm does not do?
Any help?
Regards
Bilal
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2007 Sep 20
2
Outgoing SIP packets out of order?
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the
PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something
strange, at least to me. All of the SIP packets going out from our Asterisk
PBX to either of our 2 VoIP providers are consistently 50% out of order. In
addition, if I use Wireshark's voip call player, the outgoing side of the
call
2008 Oct 29
4
Dimensioning a telephony system based on openser!
Hi,
I've sucessfully completed an Openser 1.3.2 + Mediaproxy 1.9.1 + Asterisk
1.4 + CDRTool with freeradius telephony system.
Asterisk is used only for voice mail and redirectioning calls.
Every calls should pass through mediaproxy so that i can account them.
The goal was to create a simple prototype of what could be a VoIP
provider.
Now i need to dimensioning this system to work
2009 Sep 08
2
1.2 AGI Deadlock
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I
get the "avoided deadlock" message below.
*CLI> == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on
'SIP/3211-1-081c40a8'
-- Executing NoOp("SIP/3211-1-081c40a8", "") in new stack
-- Executing AGI("SIP/3211-1-081c40a8", "diallocal.agi") in new
2008 Jul 09
2
Asterisk dimensioning
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
asterisk machine?
Which is the best way to install that? two asterisk with openser. One
asterisk with openser .....
Is it necesary run a SER server on this enviroment?
Any clue will be welcomed.
Thanks in advance.
VoipCrazy
2008 Jul 22
8
Cisco vs Asterisk
Hello all,
A client of us, is thinking to migrate their actual PBX to a Cisco
CallManager. We want to sell him an asterisk box to complement the
Cisco PBX.
I think to use asterisk as a Voicemail server (Replazing the Cisco Unity)
Has asterisk all the functionalities to replace a CIsco Unity server?
Which functionalities Cisco Unity has than asterisk could cover?
How could asterisk complement the
2011 Nov 27
6
Does Asterisk alter the Headers of INVITE Message
Hi all,
I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there
So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message?
Thank u
2007 Jul 30
5
Silly MeetMe() question.
I've got the ztdummy kernel module loaded and seem to have all the desired
prerequisites in place, but Asterisk never seems to compile with MeetMe()
application support enabled, nor does there appear to be a module I am
failing to load that would contain this application.
Is there something really obvious I am missing?
Thanks,
--
Alex Balashov
Evariste Systems
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