are used in those cases.
Saddly enough, i still didnt find the time to do any load measurements
on pri cards. Although i have a test setup ready to go.
Zoa.
Eric Giesselbach wrote:
Hi,
During tests with a IAX2/PSTN gateway I've been getting strange results for
processor idle time and load. I (re
),
you can find some of the results on www.astertest.com (there you will
find also some imature version of a callgenerator for asterisk that
would probably help you to do things faster).
I could also help you off list if you want.
Zoa.
Kristian Kielhofner wrote:
Hello,
I would like to test
Its a very very bad idea to do this on production boxes. Especially if
you are trying to see how far you can go, and then you cross that tiny
border :)
Your production calls will not like an idle cpu% of 0% and a load of 500.
zoa,
Bicom Systems wrote:
[EMAIL PROTECTED] wrote:
Hey,
Most
Its the most crappy piece of software i've ever seen.
Not to mention that even the installation files for the latest version
are incomplete. (try importing the .sql files).
Joachim.
Kanuri, Seshu (Company IT) wrote:
it is by far the most complete billing system.
Really? Are you using it now or
cpu load on te4xxp cards is very low, and now that they have echo
cancellers as add-ons cards, it will be even lower.
I can't speak on hardware compatibility as i never tried a sangoma card.
(But i can say that in the last year i've never had an issue with digium
cards and we have 8 in use.) The
I didnt have to do a single restart in about 2 million calls on te4xpp
so far.
David Brodbeck wrote:
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the
There is a small forum on www.astertest.com
All not stupid questions will be answered there, or at least we will try
to give an answer. (e.g. all questions where the answer doesnt show up
on page 1 when you type the question in google).
/Z.
Sean A. Newton wrote:
On Thu, 31 Mar 2005, Chuck Bunn
Yes they do, it was presented at von. Its a little daughterboard for
te4xxp cards.
Jerry wrote:
Digium has a hardware echo can?
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Description: OpenPGP digital signature
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Hi,
Could you please include if you used the software zaptel echo canceller
or the daughterboards for the te4xxp ?
As that would explain the difference in cpu usage.
I have no daughterboards for the te410p cards yet, nor do i own any
sangoma things, so no testing here.
Joachim.
Joachim.
Tom wrote:
Sounds like a message from mpg123,
Is your asterisk crashing when this happens or are you just giving the
wrong input files for moh ?
Zoa.
amna saleem wrote:
hi!
I wanted to ask if someone ever got the error flexible rate not heavily tested
I am not able to dial from PSTN to iaxphones(on which
It would be very much appreciated if you would send a diff -u of those
files to the bugs.digium.com website for possible inclusion into a version
0.9.0
Don't forget to send a disclaimer.
Joachim
At 07:07 16/01/2004 -0800, you wrote:
I too would like this.
Thanks so much
~paul
This is absolutely not true.
I have 3 (raid) scsi asterisk machines in production.
Joachim.
At 11:32 21/01/2004 -0500, you wrote:
In my view at least one IDE drive must be installed in order for * g729
license to work.
To simplyfy, here is the matrix (This is how I think it is please
confirm)
have a choice, i'd go for ilbc, sound quality is better, packetloss
features are great
At 22:28 21/01/2004 +, you wrote:
zoa wrote:
This is absolutely not true.
I have 3 (raid) scsi asterisk machines in production.
Joachim.
At 11:32 21/01/2004 -0500, you wrote:
In my view at least
I had a similar problem a while ago,
The g729 negotiation with chan_h323 might cause problems sometimes with
compatibility between g729a and g729b.
While g729a and b are perfectly compatible, the as5300 might have problems
recognizing g729b as g729.
(I had to allow g729a,b and ab on my hardware
Zoiper can do it when you use the provisioning, contact me offlist on
[EMAIL PROTECTED]
Zoa
Joao Pereira wrote:
I don't think so, because in paging/intercom, the phones must support
Auto Answer.
The link you sent says:
SIP phones for the most part don't support any of these phone based
Mexuar is the best known one i think, they showed me a demo on
astridevcon, seemed to work ok.
Zoa
Matthew Rubenstein wrote:
Does anyone know of an IAX softphone in Java, whether applet or
application? Even the most minimum featureset, just voice and dialing,
or even embedded
Same here
lenz wrote:
Mee too, a lot of the messages I'm sending seem to disappear.
l.
In data Tue, 02 Oct 2007 22:38:26 +0200, robert boardman
[EMAIL PROTECTED] ha scritto:
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to
Use the astmanproxy and move the load elsewhere. (If you just want to
passively listen to messages, your box is about 100 times faster than
you need :)
Zoa
Roberto wrote:
Have anyone maided like 200 simultaneous connections to asterisk AMI
(manager). ??
How many connections can I
I would stay with DECT, the battery in WIFI devices only lasts a couple
of hours. (Unless you want to take the phone with you and use it on
public hotspots etc)
Zoa
Luis Antonio Prata Barbosa wrote:
Some days ago, I was looking for some mobility solutions...
My conclusion is Wi-Fi
to be the same.
Regards,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of zoa
Sent: Friday, November 21, 2003 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
(Alpha)
Could you give me
compiler ? Any
speed gains ?
I'm also very interested in anything that could give me a speed gain,
compiler settings, kernel tweaking, etc etc... let me know :)
zoa.
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run ldconfig, reboot the server and normally all will be fine, if not you
will have to reregister.
I've seen it before, and i'm sure you will see it again :-p
At 16:20 22/11/2003 -0500, you wrote:
Hey all,
Does anyone know what this means?
I was running asterisk fine. Installed it on
i think that is a bad idea, atm i have the option to use my screen speakers
for ringing and my headset for the actual audio. My pc speaker sux bigtime
(too quiet) but i agree that putting an option for the pc speaker is a good
idea.
Zoa.
Ok... you're right. I'll make it to take the default
I am having some issues when trying to connect with perl to the asterisk
manager and doing an IAX2 show channels.
If i do that on a server that is heavily loaded, i sometimes get some
events instead of the channels i asked for.
Any suggestions how i could fix that ?
zoa
I am a bit puzzled about the meaning of the different jitter buffer options.
If i set:
Dropcount=3 what effect will this have exactly ? (will this have an
influence on how fast a jitter buffer is built or destroyed?)
jitterbuffer=200 - will this create a fixed buffer of 200ms this implying
Count me and one of my collegue's in.
How long are you staying in Paris ? The 19th might be a bit early for us,
but then again maybe not :)
Zoa.
At 23:28 29/11/2003 -0600, you wrote:
I'm coming to Paris Dec 19. I was wondering if there was any interest in
having an Asterisk get together
And while you are in Europe, why not also do Brussels ? ;)
zoa.
At 11:16 1/12/2003 +, you wrote:
Mark,
We're happy to host something in London if you were dropping round these
sides.
Tan
Telappliant.com
Voiptalk.org
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Mini meeting next week lubomir ?
i'll be there starting on monday :)
zoa.
At 16:00 1/12/2003 +0200, you wrote:
Sofia (Bulgaria) !!! :)))
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam
anyone with some succes with the drivers for the TE410p ?
joachim.
At 14:06 6/12/2003 +, you wrote:
Hi ,
I picked up a x100p the other day and thaught I'd havea go at getting the
driver going for linux 2.6, things have gone pretty, two basic problems.
1. makefiles, with 2.6 you can't
I seem to have the same problem now,
were you able to resolve this ?
joachim.
At 22:41 6/11/2003 -0500, you wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot
try running ztcfg first
At 09:26 29/12/2003 -0500, you wrote:
Hello all
I just checked out the latest
zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through
the entire make procedures. Everything seemed to go fine however now when
I attempt to start asterisk, it says ok but
You also don't need such a complicated perl script, just muxing them
without cutting them is enough.
(Timing was fixed)
zoa.
At 14:41 4/01/2004 -0600, you wrote:
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote
mail em to the list.
Cheers,
Zoa.
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We are considering it yes, but i don't know how hard or easy it would be.
I guess we will first try to make the other versions like we want them
to be and then start looking at other os'es.
Zoa
Aldo Bergamini wrote:
[EMAIL PROTECTED] is believed to have said:
Hey ho,
A few days ago
Google is your friend, from http://www.voip-info.org/wiki-VOIP+Phones :
http://iaxclient.sourceforge.net/iaxcomm/index.html
http://www.asteriskguru.com/tools/idefisk_beta.php
http://www.voip-info.org/wiki/view/GnoPhone
Giorgio Incantalupo wrote:
Hi,
does anybody know if there's a iax phone
All versions released in the last 2 weeks i think.
take the newest versions from svn or the ftp. (1.2.3 is released).
Cheers,
Zoa
Darren Ellis wrote:
Olle E Johansson wrote:
This morning we discovered a serious bug that stopped all bridged
audio in our Asterisk servers. Mark found
,
and how fast your server is.
Zoa
Andrew Nowrot wrote:
Hi,
Does anyone know what is the amount of max concurrent calls that can
be made in one Asterisk box?
I heard that it is 256 and it doesn't depend on how good your machine
is. It is the program constraint. What can I do when I need to have
If you really need it, save it on a remote server (nfs or so), that
should minimize the problems
Zoa
---
www.asteriskguru.com
Simone Cittadini wrote:
Moises Silva ha scritto:
How important is the impact i could have if I have a single entry log
file in /etc/asterisk/logger.conf wich loggs
It can be done, are those 3000 calls sip to sip ? If so it could easily
be done, if they are not sip to sip you will need a bunch of servers.
Zoa.
Vic wrote:
Hi,
we are currently considering different options for rolling out a large
scale IP PBX to handle around 3,000 + concurrent calls
over 100 calls).
Its a very long time ago, maybe some of those issues are resolved by now.
Zoa
trixter aka Bret McDanel wrote:
I am curious if anyone has had problems trunking iax2 with 100+
concurrent calls. I am planning on testing this out tomorrow, however I
wanted to know if anyone else
in smaller trunks.
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 13:38 +0200, Zoa wrote:
A long time ago i tried to make one big iax2 trunk for one of my
customers, i soon changed this to several small trunks. (bandwith doesnt
rise all that much if you use 2 trunks instead of 1
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
I think
When you have a lot of calls, try doing a show channels and iax2 trunk
debug. (they are killers)
Zoa
trixter aka Bret McDanel wrote:
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do
Any Asterisk people going to Cebit ?
Let's meet! If you go and would like to go for a drink and meet some
others from the voip business, please add your name to the list below
Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com -
wednesday / thursday.
Tan Aksoy - Telappliant -
So far these people let me know there are going to be there, who else is
going and wants to do some networking
Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com -
wednesday / thursday.
Tan Aksoy - Telappliant - wednesday / thursday
Antoine Megalla - SAND - wednesday /
a built in switch ?
- How many lines will your agents handle ?
- do you need busy lamp fields
- do they need to be provisioned through tftp ?:
Zoa
Mail list wrote:
Hello
Can anyone suggest sip phones with headset for use in call centers .
They should be fully inter operable with Asterisk
Mojo with Horan Company, LLC wrote:
[EMAIL PROTECTED] wrote:
I am planning to write a module to find if a Special Information was
detected or not.
Can anyone please help me to figure out the below fields?
1. The Frequency of a frame
2. Length of frame in milliseconds
Contact me at [EMAIL PROTECTED] and ask for a beta for the 64 bit build
of zoiper
Cheers,
Zoa
martin f krafft wrote:
Hi,
I am on amd64 Linux and not really too happy with twinkle, linphone
and ekiga. Unfortunately, X-Lite and Zoiper, even though they
provide Linux versions (w00t!) have
the reregistration time is set on those end
devices and how much the registrations will collide in the same small
interval.
SER doesn't handle audio so even if the registration gets a little
delayed because a flood arrives, the audio won't suffer.
Zoa
Abid Saleem Choudhary wrote:
Hi All,
I
I'd say, save yourself the time and the frustration, drop the idea and
buy a real voice card.
Zoa
Ronny Forberger wrote:
Hi,
maybe this has been asked before but I couldnt find a proper answer on
the web or list.
I want to use a analog V.92 modem to make outgoing (and possibly
If you cant power off the machine, look for a sip ata or channel bank.
USB/ TDMoE Channel banks:
xorcom.com
spidermux.com/
And for ata's or sip gateways, there are zillions of brands,
Zoa
Ronny Forberger wrote:
Thanks for that. What channel module do I have to use then ?
And can you
Looks like a standard chatbox with flash media server in between.
You can't use this with asterisk unless you write a flash media server
channel or a convertor of some kind.
Zoa
Dean Collins wrote:
Interesting to note that Tokbox now has ‘clientless’ voice and video
conferencing
modules, lumenvox does it for voice recognition,... ).
OT, where can i find the best info on this salesforce API ? Do you see
any possibilities to integrate our zoiper softphone with salesforce ?
(contact me off list for that)
Cheers,
Zoa
Dean Collins wrote:
Hi BJ,
Further explanation about
How about a tail -f on Master.csv ?
Then you will have everything realtime and you will not need a cronjob.
Zoa
Col Ferguson wrote:
Hello again,
I can copy the file out the serial port by doing this:
rename Master.csv out1.csv
cat out1.csv /dev/ttyS0
If I build a script to do
What is app_swift ?
Zoa
Darren Sessions wrote:
Thought I'd let everyone know I've released app_swift v1.6.1 which is
entirely based off of Will Orton's work he's placed in the public
domain.
Works great with Asterisk v1.6.0-beta7.1.
In any case, can be downloaded from my site at:
http
Afaik its per encode / decoder pair.
In this case you will need 32 simultaneous encoders / decoders between
g729 and slin, so you would need 32 licenses.
Contact digium sales/support directly and you will know for sure :)
Zoa
Carlos Chavez wrote:
I need a refresher course on how many
You might need to set the dialplan to international or so in the config
files.
Zoa
Stuart Ford wrote:
Hello all
As always I'm trying the mailing list as a last resort as I'm out of
options. I am seemingly unable to dial international numbers over our BT
ISDN30 line.
I've checked
congratz!
Zoa
John Todd wrote:
On Tue, May 20, 2008 at 7:41 PM, John Todd [EMAIL PROTECTED] wrote:
I'd like to take a few moments to introduce myself and the new role
Hi John,
Like Jared, you need no introduction to most of us, you are a pillar
of the asterisk community
I think you can buy some kind of ATA's to do such a job.
I do not however remember any brand names
Google returned these links:
http://www.voip-info.org/wiki/view/Asterisk+phone+doorview_comment_id=15775
http://www.abptech.com/products/its.html
Mobotix
c james wrote:
I have an opportunity to
if it did it would be a pita to set up).
I think you would be better of writing a script that generates call files.
Zoa
Sebastien Cruaux wrote:
Hi,
Did someone ever managed to make Astertest
(http://www.asteriskguru.com/tutorials/astertest.html) work ? I
followed all the instructions
it a try.
Cheers,
Zoa
James FitzGibbon wrote:
Has anyone found a softphone that supports pulling it's configuration
from a central server via TFTP/FTP/HTTP, much like hard desk phones use?
I'm looking for something for a call center that I can provision from
a central location by generating config
For things running inside the browser, i think java is a reasonable
choice. Yes you could do it with active-x too, but it won't work on all
OS'es. I hate java, probably for the same reasons you do, but in same
cases its the best option.
Zoa
Dean Collins wrote:
Lol, yep you missed
Several people do use it for handling 50k minutes a day. (I'm one of
them).
Yes, you need to know what you are doing, and have a nice design, but it
is possible.Our code is only slightly altered. (mainly for billing
purposes).
Zoa
Daryl Jurbala wrote:
On May 12, 2007, at 4:11 PM
Gordon Henderson wrote:
On Sun, 3 Jun 2007, Andrew Kohlsmith wrote:
On Sunday 03 June 2007 4:30 pm, Alex Crow wrote:
No frills, specs look good, price seems excellent!
http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519
That's a terrible phone. I've tried them. the screen
We have it (in belgium)
http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-sl75-wlan-voip-phone.html
I still think DECT is better though :)
Zoa
Alex Crow wrote:
Alban,
Thanks! Where on earth did you source this? I can't seen to find hide
nor hair of it here in the UK :(
Alex
Return the card and ask for a new one. (i have seen this problem before
with a broken 411, a new card fixed it).
Zoa.
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List
Just want to check if anybody else is having this problem
Allison is not exclusively working for asterisk, she also does other
recordings.
Zao
Steve Totaro wrote:
Just got a call from Ebay's unwired buyer and The Voice is Allison
Smith.
Adoption is wide but who is willing to give away their competitive
edge (although ebay doesn't really have any
Hello,
Send an email to [EMAIL PROTECTED] i think we the upcoming
version has some fix for this iirc
Zoa
Nir Simionovich wrote:
Hi Philipp,
Thanks for the tip, but that is not what I initially meant. I'm
using IDEfisk, and I would like it when a call comes
Into IDEfisk to generate
So does asterisk (Albeit with a commercial package)
http://www.attractel.com/t38.html
Lee Howard wrote:
Matt Riddell [NZ] wrote:
Does OpenPBX do a T.38 gateway then?
Yes, it does.
Lee.
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I wouldn't do that with softphones, unless the softphones are designed
to do this.
The delay will vary depending on the audio card, OS, and drivers.
(And the phones might not all answer at the same time, but if you use
music on hold or so to play that should not be a problem).
[EMAIL
registered to Asterisk A, it - at least
used to - not work very well in production).
If you do a lot of simultaneous calls, make sure your vpn servers can
handle the load.
Zoa
www.asteriskguru.com
Michelle Dupuis wrote:
You will likely have latency issues - causing choppiness. Start
You could use an rebranded (OEM) idefisk - does sip and IAX and uses
XML for the config files, not the registry - making it possible to use
it on a usb stick.
More info : http://www.asteriskguru.com/idefisk/oem/
(But its not open source, nor free).
Joachim
Mike Lynchfield wrote:
sip
Joe Acquisto wrote:
. . .
http://sourceforge.net/projects/stun/
Which is linked from:
http://www.vovida.org/applications/downloads/stun/
That's what I'm running.
Gordon
Thanks. Looking there, why would I need a stun client if the
device/softdevice already has STUN
You can find cheap gsm (+/- 150$) gateways too, although the cheap ones
will require a additional pstn card. (expensive ones could do sip)
Zoa.
Jay Milk wrote:
There's been some (futile?) effort a while back attempting to get a
Bluetooth capable phone integrated into asterisk as a channel
Check the timer frequency, it might have a different setting on the two
kernels.
RR wrote:
Hi all, (2nd attempt)
this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with
to reproduce it on command.
I have several other te410p's on different locations (with different
carriers), without those complaints.
Does this also happen on pri to pri calls for you ?
Maybe its a combination of carrier volume with the te410p ?
Zoa
Servetas, Andrew wrote:
We are experiencing
But does it help ? Is it better than before ?
Do you have a good way of debugging ? (like an audio recording that i
could play ?)
Does it show something on the cli when it happens ?
Zoa
Servetas, Andrew wrote:
They recommended changing the default value of 1000 up or down
incrementally
- imap storage for voicemail
- whisper paging
- Autoconf configuration
- menuselect (graphical module select tool similar to the kernel config
system)
- higher quality prompts (in English, French and Spanish). - watch out
they are restructured a little
Zoa.
Roy Sigurd Karlsbakk wrote:
Hi all
quad port T1 card
3 channel banks.
Zoa
mike wrote:
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
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asterisk
linksys spa3102 or 2100 are known to work.
Grandstream also should do it with recent firmware.
Don't be fooled by what is written on the box, lot of ata's out there
claim t.38. (while the firmware doesnt contain anything related to t.38)
Zoa
Christopher Corn wrote:
lee,
Thanks
I can confirm the same.
It doesnt mean the audio will be delayed, the phone is just slow with
replying to the sip messages.
Zoa
Michiel van Baak wrote:
On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm sure other people
/pci_irq_apic_tdm_ticks_te410p_te405p_noise.html
and search for _Put your networkcard and pri card on a different CPU
Zoa.
_
Robert Jenkins wrote:
Hi,
On Centos IRQBalance should already be available.
You should be able to run 'setup' from a console/terminal, go to System
Services enable irqbalance
Looks like phonality has bought trixbox. (I suppose they failed to buy
digium :)
http://news.asteriskguru.com/10/773/2006/10/5/Fonality_Aquires_trixbox_([EMAIL
PROTECTED])
Earlier on they found venture capitalist:
http://www.fonality.com/press/20060109.htm
know any 1 port cards that do NT mode.
Zoa
Ejay Hire wrote:
Hi. A cross-over cable won't work, the isdn network provides signalling
and adressing functions.
When I was studying for my CCIE, an ISDN simulator cost an arm-and-a-leg,
around $1k used from ebay.
-ejay
-Original Message
I will be also on a flight from frankfurt (lufthansa), but a few days early.
Zoa.
Stelios Koroneos wrote:
Greetings !
Its kind of OT, but if there are any Europeans going to Astricon in Dallas,
please send a message of-list.
It's possible we will be on the same flight,(i am flying from
Yes
Akpome Akpoguma wrote:
Hi All,
Would asterisk and zaptel compile on 64bit dual xeon hardware??
Rgds
From: Martin Joseph [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject:
Xlite is not GPL!
Joe Dennick wrote:
The X-Ten is probably the most know free soft-phone availible. You
can find it at
http://www.xten.com/index.php?menu=Productssmenu=xlite
Gregory Duchatelet wrote:
Hi,
I’m searching for GPLed softphones. I found WengoPhone but actually
not available
Digium sells cables to interconnect them for timing. (dunno if thats
only for the 412 cards).
zoa
Don wrote:
As long as you have no interrupt conflicts...don't see why not...
We have 3 TE410P cards in a Dell 2850...had to disable hyperthreading
in the bios...and then make sure we had
From our experience, chan_jabber doesnt work behind nat. We tried to
patch it (in a similar way as nat=yes in chan_sip) but quickly bumped
into other problems.
(problems explained on mantis).
Zoa.
Gustavo Hernandez Baratta wrote:
Hi!
I´m trying with 1.4b2, chan_jabber and chan_gtalk
I have such a setup here myself, although not for 100 people.
Any recent server will do, but make sure you don't call 100 people the
same second, spread them a little over time.
Google for .call files
Zoa.
Ady Wicaksono wrote:
Imagine i want to create application like SMS Alert, however it's
Yes, its the same as what we tried.
Gustavo Hernandez Baratta wrote:
Hi Zoa:
Thanks for your answer. Let me explain: Asterisk are not behind a NAT,
google talk user are. Do you think that is the same problem?
Thanks a lot!
gus
At 10:28 a.m. 17/10/2006, you wrote:
From our experience
Lets change the question to : does somebody know good iax phones, that
are ROHS compliant and without enormous delivery problems ?
Neil Tancock wrote:
Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution and my SIP phones are good but not
It's not possible.
The idefisk however has a button to auto answer.
Zoa
Gregory Duchatelet wrote:
Hi list,
I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to
Asterisk. One call the other-one, is it possible to order Asterisk to
force answering the call ? i.e. Xlite call
get exceptional support or they ship me handpicked gold plated,
overclocked versions of their cards (not really since i just buy them
from a reseller).
Cheers,
Zoa.
Dovid B wrote:
Can I now 5th it ? All this makes me wonder why Digium dosent work
harder. I have mainly only seen others praise
Those 2 softphones below only do SIP, no IAX.
Zoa,
--
www.asteriskguru.com
Mohamed A. Gombolaty wrote:
Hi Wasim,
Check out the x-lite softphone
http://www.xten.com/
As for linux check this page there are two softphones type available :
http://www.iptel.org/products
Thx
MAG
wassim
Could someone also tell me how much a car costs ?
What i mean is, it all depends on your server and the codecs used, the
max is currently a DS3 worth of calls.
altus wrote:
Good day all
What is the amount of calls that asterisk can handle,SIP and from/to
PSTN
signature.asc
The bandwidth calculator at
http://www.asteriskguru.com/tools/bandwidth_calculator.php
will tell you exactly how much bandwidth your calls will take, in both
directions.
Zoa.
Karlheinz Hagen wrote:
How can I calculate the quantity of concurrent users using a
bandwith of 512Kbps ?
http
Or you could get it (or at least something similar) for free from
www.asteriskguru.com. A small preview is available here:
http://www.asteriskguru.com/tutorials/queue_stats.html
Its 100% ready, just waiting to be uploaded. (Should be there in the
next few days).
Zoa.
William Boehlke wrote
Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but
line is in PRI Signalling signalling
This is your problem, probably in one configuration file you have fxo
kewlstart as signalling, and in the other one you have pri signalling.
Greetz,
Zoa
---
http://www.asteriskguru.com
As promised, we just released the first version of the asteriskguru
Queue Statistics.
Screenshots and download at:
http://www.asteriskguru.com/tools/queue_stats.php
---
Small description:
The Asteriskguru queue statistics, is a PHP based program, which gives
anyone who uses queueing in
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