?page=Asterisk+cmd+VoiceMail
http://www.voip-info.org/wiki/view/Asterisk+cmd+Background
Trying to do that from the start might be a bit fast, i'd suggest
looking for a beginner tutorial first.
Greetings,
Zoa
On 9/7/2010 4:16 PM, Frenette, Rob wrote:
Hi,
Does Asterisk, provide the option within
Colin,
I'm working for Zoiper, you can contact us directly on supp...@zoiper.com
Zoa
Nick Brown wrote:
Do you see the issue when calling between two softphones? Do you see the
issue if you call from your mobile into an echo test?
Setting TOS flags on packets will make no difference
if
needed) and see if the problem goes away.
Greetings,
zOa
Vieri wrote:
--- On Thu, 5/13/10, Zoa zoach...@securax.org wrote:
Can you try trunk = no ?
Lifesaver...
trunk=no made the interference go away.
I have clean audio now.
Quote: IAX Trunking needs support of a hardware timer
Hello,
Can you try trunk = no ?
How much jitter do you see on the link ?
Zoa
Gareth Blades wrote:
There should be no noticeable difference between slin, ulaw and alaw so
what you have is fine. The problem must be elsewhere.
Vieri wrote:
--- On Thu, 5/13/10, Gareth Blades list-aster
I have played with one before, it worked quite well. (Until somebody
fried it by accident).
Joachim
Peter wrote:
Hi,
I have one in stock - got it from a client who wanted to get rid of all
his old IT equipment.
Looks strange, did not have enough time to play with it Tried it
once,
JR Richardson wrote:
Zoa wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent gatewaying.
The solution should be quite stable as long as the line quality is ok.
(Some tools for measuring the line quality
Thanks,
I have uploaded the patch to the website and will let you know the
feedback we receive.
Greetings,
Joachim
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New to Asterisk? Join us for a
at 4:52 AM, Zoa zoach...@securax.org
mailto:zoach...@securax.org wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent
gatewaying.
The solution should be quite stable as long
as well).
Greetings,
Zoa
On 3/7/2010 1:50 AM, Thorolf Godawa wrote:
Hi,
I am looking for an Mail-2-Fax and in a second step Fax-2-Mail-solution
that works via T38 with Asterisk, currently still version 1.4 but it
also should work with 1.6.
For Mail-2-Fax I am thinking that you either have
/opensource_fax_stack_PR.pdf the project
homepage can be found at www.zoiper.com/foip/
Cheers,
Zoa
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New to Asterisk? Join us for a live introductory webinar every Thurs
Hello,
Give our zoiper softphones a try, you could achieve this functionality
by sending a url over IAX (Sendurl) or by using the open website on
incoming call. (In which you pass the callerid as a paramter to the
website to open the ticket that matches that one. (You could also ask
for the
Zoiper supports it in our wideband beta:
http://www.zoiper.com/downloads/beta/Zoiper%20Communicator_Free_1.12wb2_Installer.exe
(the beta is a bit old though).
Cheers,
Zoa
On 2/20/2010 8:02 PM, Kyle Kienapfel wrote:
Hi, I stumbled upon mentions of a SILK codec last night on skypes
skype
I have seen this years ago, i received complaints about women voices
triggering dtmf.
With some help from Mr. Underwood, it was able to confirm lots of false
positives on the dtmf detection.
My issues went away when we upgraded all cards to the ones with the
octasic DSP chip on them.
Zoa
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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 /
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 -
Have a look at our Zoiper (http://www.zoiper.com/oem.php) - it does all
4 items you are looking for.
Zoa
Dovid B wrote:
Try EyeBeam. It is the paid version of X-Lite.
- Original Message -
*From:* Mike mailto:[EMAIL PROTECTED]
*To:* 'Asterisk Users Mailing List - Non
T.38 will not work with the fxo card.
Zoa
Fernando Berretta wrote:
Dear All,
Are you telling me Asterisk 1.6.0b2/4 has support for t38 and rxfax
etc. and will be able to receive faxes and negotiate with voip CPE's
like ATA's to transmit faxes which comes from FXO cards to VoIP
Devices
Fernando Berretta wrote:
Tzafir,
I'm sorry, my question wasn't clear.
Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some
modifications on app_fax so the questions are:
1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO
Card and this FXO port is
part.
Zoa.
Rob Hillis wrote:
T.38 is a codec in exactly the same way that GSM or G.729 is a codec,
so yes it /can/ be used at the same time as any other codec - just
that only /one/ codec will be used at a time. What often happens is
that the call will initially be established
Asterisk does not support that yet.
Zoa
rachid wrote:
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I
I'm working for zoiper.com and i'm willing to help out with ours when
needed.
Zoa
d4rk f1br wrote:
Anyone aware of any SIP softphones that might virtualize well with
Citrix presentation server? I suspect I know the answer already as I
have been researching softphones that work
Iirc, there used to be such an adaptor in the digium dev kit years ago.
Maybe somebody here remembers what it was exactly ?
Zoa
John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the
place to get the best answer. I have looked at the many Skype
Gordon Henderson wrote:
On Sun, 27 Jan 2008, John Millican wrote:
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at
is the command line:
tar zxf zoiper201-linux.tar.gz
./zoiper
3) Start Zoiper.
*ZoIPer depends on ALSA library, so it* **must** *be installed!
*
Zoa
Robert Moskowitz wrote:
zoa wrote:
Have you tried our Zoiper softphone yet (www.zoiper.com) - new
version scheduled for in a couple of days ? If so
://www.zoiper.com/biz3.php
I have an example for jscript somewhere tool, contact me offlist if you
want it. Let me know offlist if you need any biz licenses to try it out,
i;d be happy to provide you with them.
Zoa.
Christian Ejlertsen wrote:
Ok good piece software easy on the eyes as they say and I have
Hello,
Have you tried our Zoiper softphone yet (www.zoiper.com) - new version
scheduled for in a couple of days ? If so, can you send me any remarks
of list so that we can keep those things in mind for future versions ?
Greetings,
Joachim
Philipp Kempgen wrote:
Andre Herrlich wrote:
I'd say check with Digium, maybe it's supposed to not break (i
personally don't think it would break it, i'd have noticed it already
:) if you plug it to the wrong thing and you will get a replacement for
free.
Zoa
bilal ghayyad wrote:
Hi All;
If one of my FXS port damaged at TDM22B
Philipp Kempgen wrote:
Siju George wrote:
What are the security ramifications of peering two Asterisk servers on
remote locations and sending the VOIP traffice through the internet
using IAX2 ? Can this traffic be sniffed and the Voice be captured and
heard by any third party?
There are many, (i'm one of the people working for zoiper):
Look at the iaxclient homepage,
There are iaxcomm, loudhush, kiax, mediax , diax and many more,
(you could also easily make your own).
Cheers,
Zoa
Vincent wrote:
On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED
IAX had some stability issues in the past, the recent releases have a
lot of iax2 fixes and should no longer have those issues.
Zoa
Steve Totaro wrote:
randulo wrote:
Hi,
We all know what the principal advantage of IAX is, doing it all on a
single port, right? But now and again I
The jitter buffer is actually the same.
Zoa
Dr. Michael J. Chudobiak wrote:
randulo wrote:
On Nov 30, 2007 1:40 PM, Steve Totaro [EMAIL PROTECTED] wrote:
solved these issues. I think trunking (one of the main selling points
of IAX due to less overhead) may be a common denominator
their first steps in the asterisk world
Crossposted to -users.
Zoa
Luigi Rizzo wrote:
As a result of the commit below, now trunk can be built and run under
Windows/cygwin, including the building of modules.
Haven't checked yet the functionality - some modules surely cause
ill side effects
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
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
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
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
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
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
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
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
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
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 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
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
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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--Bandwidth and Colocation provided by Easynews.com --
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
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
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
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
You need a timing device on both ends.
Zoa
Vicky wrote:
If the other server doesnt have any hardware device that can act as
timer. then just compile zaptel and modprobe ztdummy .. This kernel
module should act as timing source i think . ( it works with meetme ) .
On 16/01/07, *Andy Hester
Does somebody know a similar device that does the same for GSM networks ?
Zoa
Dovid B wrote:
There has been talk about it before and I think people have done it.
Paging Sam Tam
- Original Message - From: Joao Pereira [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non
Yes
Zoa
Michel wrote:
Hello,
How many licenses to buy?? :
From what we understood from digium website, we must buy as many
licenses as the number of maximum simultaneous calls using G729 Codec
we wish to make.
For example, If we want to be able to make a maximum of 10
simultaneous
I did some tests a long time ago and the speed was roughly the same. ( I
think digium's was slightly faster).
I think the IPP version also doesn't work on AMD out of the box.
It's just 10$ a channel, that's not even worth the hassle of trying
something else.
Joachim
Al Bochter wrote:
It used to be a problem to have very big iax2 trunks (e.g. 100 channels).
This should be resolved in asterisk 1.4, in older versions you can just
work around it by making several smaller trunks.
Zoa
Noah Miller wrote:
Hi Adrian -
(Happy new year!)
How big can an IAX channel grow
Have a look at www.spidermux.org
Zoa
Allen Casteran wrote:
We have an application for Asterisk that will require connecting 144
fax ports into the system. Faxes will route externally over a PRI. The
144 ports are for local fax machines within the building. Not all will
be faxing
Idefisk will do that - www.asteriskguru.com . (And asterisk will accept it).
Zoa
Al Bochter wrote:
Ok does anyone know of any softphones that will dial DTMF tone keys A
B C D
And do you know if Asterisk will take the DTMF Tones for A B C D
Hmm, if the latest free version does not have all 16 keys, email
[EMAIL PROTECTED], there should not be a difference in the amount
of DTMF keys between biz and free version.
Zoa
Bob Chiodini wrote:
The free version 1.31 has all 16 keys on the keypad.
Bob...
Al Bochter wrote:
Are you
Idefisk 2.0 will have it.
Zoa
Mail list wrote:
Is there any good iax2 softphone capable of attended transfer ( like
sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to
handle attended transfers
It is scheduled for 9 januari. (If you ask nicely on
[EMAIL PROTECTED] and promise to give good feedback, you might be
able to get a beta version earlier ;)
Zoa
Vicky wrote:
I have configure it by using the *2 atxfer feature of asterisk but its
not as good as other attended transfer which
You could go for 2 quad pri cards + channel banks or for TDMoE or usb
channel banks.
The last option would be the cheaper and more scalable one imho
www.spidermux.org
www.xorcom.com
Joachim
John covici wrote:
You could put at least two Rhino quad t1 cards and that would give you
8 times 24
with recent kernels). I know the spidermux people already have a
bunch of patches ready to be released to fix the issues that exist now.
I've never heard something about tdmoe being phased out of asterisk.
Zoa.
___
--Bandwidth and Colocation provided
http://www.voipsolutions.be/phones/dect-phones/gigaset-sl75-wlan.html
Zoa
Andrew Joakimsen wrote:
Where can it be purchase?
On 11/21/06, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
yes I tested this one week ago
Can you tell us how you do the testing ?
Zoa.
Anton Tinchev wrote:
Anybody sucessfully got stable 1000Hz clock without Digium harware and
kernel 2.6?
We need to consult some peoples how to clock asterisk stable with
exactly 1000 Hz without much kernel/drives patching/tweaking.
Some test
Whatever suits you. (and whatever suits your internet provider that
might have some ports filtered without telling you).
10.000 to 20.000 is the default range afaik.
zoa
Matt wrote:
Ok,
So what exactly is the RTP port range support to be? Lots of people
are claiming 1,000 - 2,000
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
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
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
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
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
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:
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
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
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
- 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
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
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
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