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 t
Could you give me some explanation on how to use the configuration file ?
I always get INVALID channels if i click on the red icon next to my name.
(Maybe i should use a numeric context or use numeric user names?)
Tool looks great, this will be a very cool asterisk addition.
zoa.
At 16:43 21/11
e 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
e asterisk on the intel c 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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[EMAIL PROTECTED]
h
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 t
Sometimes when people hang up, or the call gets interrupted for some
reason, music on hold starts playing. (i use app_dial without extra
parameters and no moh is set in extensions.conf)
Any suggestions on how i could get rid of this 'feature' ?
gr
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 t
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 t
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:[
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 ? ;
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 ge
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 fi
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
pong
At 08:10 31/12/2003 +0800, you wrote:
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http://li
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
mail em to the list.
Cheers,
Zoa.
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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
-Origina
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)
ID
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
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
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:
>
>
s 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
://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 an
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 m
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
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 softph
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
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 establ
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 com
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
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:* 'A
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 - w
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 / thu
to have 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
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
t 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
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 ou
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 ?
>
&g
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
> confe
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 expla
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
>
> I
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 an
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 m
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
&
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 m
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 em
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
>>
>
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). ??
>
>
>
>
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
make 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
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
>> s
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
>>
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 <[EMAI
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?
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 is
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 :(
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 thi
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
lly to make it
work 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 a
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
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 fo
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
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
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 to
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:
Matt
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 Use
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
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
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 gener
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 PROTEC
t to an asterisk B 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 choppin
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 nee
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+door&view_comment_id=15775
http://www.abptech.com/products/its.html
Mobotix
c james wrote:
> I have an opportunity to
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 would
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 suppor
n
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 of
give
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
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
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
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
On
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 ni
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 c
ns 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 a
/opensource_fax_stack_PR.pdf the project
homepage can be found at www.zoiper.com/foip/
Cheers,
Zoa
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t; well at t38 gatewaying.
>
> --
> Matt
>
> On Sun, Mar 7, 2010 at 4:52 AM, Zoa <mailto:zoach...@securax.org>> wrote:
>
>
> On friday we finally released Attrafax under a GPL2 license.
> It comes with its own set of modems an
Thanks,
I have uploaded the patch to the website and will let you know the
feedback we receive.
Greetings,
Joachim
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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.
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
> o
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 iaxphone
http://www.asteriskguru.com/xlite.html
/Z
Vaniah Voip wrote:
Vamsi Pottangi wrote:
It would be easier if you could get send us your sip.conf entry and
confiuration made in x-lite
Also, please let us know where exactly the problem is. Is it
while registering the x-lite or during the call and the ex
No changes were made to chan_sip when the iax2 jitter buffer was added.
However, ive seen and hear several complaints about coredumps,
deadlocks in cvs-head chan_sip recently.
/Z
Luki wrote:
I've been running a version of the CVS without issue until
late last week when suddenly Asterisk would rand
Go have a look at http://www.asteriskguru.com/tool2.php and calculate it
for yourself.
This is without the signalling frames for call setup / teardown.
(bandwidth used by those is very small).
Greetz,
/Z
Incoming Bandwidth
Outgoing Bandwidth
Calls: 16 Calls: 16
RTP: 2.34 Kbps RTP
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