I agree that the original commercial model used for VXML gets in the way
of it's own success. It's also focussed way too much on ASR/TTS. I think
we can all agree this technology is still a future promise, even after
10+ years. But technically VXML is an interesting concept, especially
together
For compliance it's needed to support it, but it's not actually needed
to make a working application. You can just use wave files like this:
form id=MainMenu
block
audio src=helloworld.wav/
/block
/form
Voxeo actually has a nice set of tutorials and reference material on it:
Hi,
There is any api command to check the status of the extension whether the
agent is in ideal or in calling .
Can any one asset me to solve the problem. Thanks in advance
--
Warm Regards,
N.Baskar
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Another vote for Teliax.
Regards,
Diego
On Tue, Apr 21, 2009 at 11:48 AM, Kenneth Shaw k...@expitrans.com wrote:
Prabhuram,
I would recommend Teliax. I've been with them for over 5 years now,
originally with Asterisk and now with Freeswitch. I don't know of any
better provider.
On Tue,
Hi Michael,
I found the problem. I had a version mismatch between mod_mamanged and my FS
installation. So this is solved, but now I am running into the next problem. I
have written a small Class that is nearly similar to the one on the wiki pages.
The source is here (VB.NET)
Imports
Definitely!
I have a question about licensing, lib_esl is not MPL, right ?
So it can be freely used in a Collectd (GPL) plugin ?
thanks,
Leon
On Apr 21, 2009, at 10:59 PM, Brian West wrote:
Are you going to donate the complete module to the collectd project?
/b
On Apr 21, 2009, at 8:52
VXML is not only interesting but required concept.
I'm not talking about VXML itself, but presence of specification. I still
remember my confusion, when initially thinking about media and call control
dialogs and interfaces I found out XML based languages such as VXML and CCXML.
Nevertheless
wow, thanks, that's very useful !
On Apr 21, 2009, at 7:51 PM, Anthony Minessale wrote:
latest trunk now has the stats you seek in sofia status profile
profilename
On Tue, Apr 21, 2009 at 8:55 AM, Brian West br...@freeswitch.org
wrote:
You forgot the as xml versions
show channels as xml
Michael Jerris wrote:
Those files in tree go with the rpm build with goes correctly to those
directories.
Thanks Mike, but I don't understand :-/
Do you mean that I should not bother compiling from SVN and download an RPM
instead (I didn't see a source listed under
Hi,
How can I send out of band data from FS. I want to exchange out of band
alphanumeric data between two FS systems. There is a facility of this kind
in Asterisk by modifying the chan_dahdi.c file and define SUPPORT_USERUSER
information. How can I do that thing in FS?
Regards,
Vin
ESL is BSD licensed so it will be actually made more restrictive by mixing
it with GPL in this case but that was to be expected.
That's why I picked BSD for the lib so it could actually be used everywhere
with no license concerns.
On Wed, Apr 22, 2009 at 4:38 AM, Leon de Rooij
Michael,
This is exactly what I am experiencing.
On Tue, Apr 21, 2009 at 8:02 PM, Michael Collins m...@freeswitch.org wrote:
Kristian,
The symptom I'm experiencing is that no matter what language I specify, it
still plays the English sound files. Is that what you're experiencing? I've
run
Hello
I'm going through the various configuration files to figure out how they
work together, and noticed the following references: sofia.conf.xml -
sip_profiles/internal.xml - public context.
1. Am I right in understanding that a context is a file in the dialplan/
subdirectory?
2. Shouldn't
The internal profile requires authentication. Every user on that
profile would use the user_context variable to override the profile
context.
Me being paranoid when I wrote the configs I set the internal profile
to public just incase you misconfigure your system then you're not
open to
http://sourceforge.net/projects/wikipdf
On Fri, Apr 17, 2009 at 5:36 PM, David Knell d...@3c.co.uk wrote:
If someone has a way to make true mirrors that support read/write
this
would be interesting.
Do it robustly, transparently and in real time and that's the problem of
distributed
Hello
I'd like to make sure I understand this correctly: It seems like all the
settings found in autoload_configs/*.conf.xml are loaded, even if their
parent module is commented out in modules.conf.xml?
For instance:
-rw-r--r-- 1 root root 575 Apr 20 20:53 python.conf.xml
But..
Yes as part of the overall xml document!
/b
On Apr 22, 2009, at 9:36 AM, Fred-145 wrote:
So does it mean that the mod_python settings in python.conf.xml will
live in
the final Freeswitch.xml file in RAM, even though mod_python itself
isn't
loaded?
Brian West
br...@freeswitch.org
--
Hi, I am getting errors while installing FreeSwitch. I have enabled almost
all the packages (except: mod_snom, mod_ldap, mod_zap and few others). Do I
need to install dependencies or will FreeSwitch automatically find from the
internet? I have done the basic dependencies installed like gcc,
I suspect you have ptlib and opal libs installed from packages. If
you wish to compile mod_opal you have to get the latest SVN trunk. I
wrote a script in build/buildopal.sh that will do this for you.
Please run from /root/
/b
On Apr 22, 2009, at 9:44 AM, technologyinspired wrote:
Hi,
i was just thinking of throwing away my spa3102 and setting my windows
machine with FS on it to conenct to the telephone line for incomming calls,
something like a calling card gateway using a FXO card.
the reason being from the time i bought the spa3102, i haven't been able to
get the caller id
I don't know of any FXO cards for windows yet.
/b
On Apr 22, 2009, at 10:16 AM, xbipin wrote:
i was just thinking of throwing away my spa3102 and setting my windows
machine with FS on it to conenct to the telephone line for incomming
calls,
something like a calling card gateway using
I've used vxml on Voxeo's system and it's really nice to work with.
Underneath it's tags is Javascript so a FreeSWITCH with a fast TraceMonkey
engine and vxml would be great. Your MRCP project would help connect things to
other ASR/TTS systems if pocketsphinx isn't good enough. Nice package.
Hopefully the sangoma should work soon. Please contact sangoma to ask
when those cards will work on windows in FreeSWITCH.
Mike
On Apr 22, 2009, at 11:20 AM, Brian West wrote:
I don't know of any FXO cards for windows yet.
/b
On Apr 22, 2009, at 10:16 AM, xbipin wrote:
i was just
Try searching or getting help from this forum.
http://forum.voxilla.com/linksys-sipura-voip-support-forum/
Look to hwittenb for further assistance.
His customer service skills are excellent.
-Original Message-
From: xbipin bi...@xbipin.com
To:
Most lkely would be sangoma products since they have a windows driver.
As for the callerid you will only be able to receive it with analog
ports. Was that not working with the line in your country?
You should be able to get the provisioning file documentation from the
reseller you purchased the
The problem, I think, is that applications must be run from the dialplan; only
API functions can be executed from the console.
Try subclassing ApiFunction or calling your AppDemo from the dialplan.
-Michael
From: freeswitch-users-boun...@lists.freeswitch.org
Can anyone tell me what would or cause the above hang-up cause? I'm
using latest svn and get loads of these above 10 Concurrent calls
Regards
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Hello
In the wiki, it is suggested that more than one profile should be used
if libsofia is the bottleneck. When using multiple profiles to handle
incoming call and each profile having an unique port, what is the best
way to redirect and distribute incoming traffic? Is there any mod in
Peter Thanks for your reply:
I did what you said but FS still with the same problem, cant found the
user.
This is my reply to registration request
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory
domain name=$${domain}
Could you add some more details to this question? I'm not sure I understand
what you mean...
Thanks,
MC
On Wed, Apr 22, 2009 at 12:58 AM, Baskar yudha2...@gmail.com wrote:
Hi,
There is any api command to check the status of the extension whether the
agent is in ideal or in calling .
Can
Can FS authenticate with multiple SIP trunk providers to provide redundancy in
the event of the loss of connectivity to one or more providers?
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Yes, That can be accomplished in many ways in your dialplan.
http://wiki.freeswitch.org/wiki/Dialplan_XML
/b
On Apr 22, 2009, at 11:46 AM, Paul wrote:
Can FS authenticate with multiple SIP trunk providers to provide
redundancy in the event of the loss of connectivity to one or more
I think he's asking if there's a way to check the status of a registered
user; if the user is idle versus in a call.
On Wed, Apr 22, 2009 at 12:38 PM, Michael Collins m...@freeswitch.orgwrote:
Could you add some more details to this question? I'm not sure I understand
what you mean...
Thanks,
Hola!
If you cannot read Spanish fluently then please disregard this message.
To those who speak and read Spanish please get the phrase_es.xml file from
the latest SVN. It is ready for you all to review. If you have questions or
ideas on how to improve the prompts then please email me off list.
$${domain} (or any other preprocessor vars) will not be expanded on an
xml_curl return.
Mike
On Apr 22, 2009, at 12:31 PM, JuanMa wrote:
Peter Thanks for your reply:
I did what you said but FS still with the same problem, cant found the
user.
This is my reply to registration request
I have the following defined:
!-- Billing Open? --
extension name=billing_open continue=true
!-- man strftime - M-F, 9AM to 5PM --
condition field=${strftime(%w)} expression=^([1-5])$/
condition field=${strftime(%H%M)}
expression=^((09|1[0-6])[0-5][0-9]|1700)$
This again? ;) Lets see if this helps!
http://wiki.freeswitch.org/wiki/Dialplan_XML#Anatomy_of_the_XML_Dialplan
The dialplan is not processed and executed line by line... its
compiled and installed into the session before it goes into execute
state.
So you can't use set on one line then use
On Tue, Apr 21, 2009 at 5:17 PM, Michael Jerris m...@jerris.com wrote:
sound_prefix?
Mike
I can find no evidence of a sound prefix issue. Here's my language setup
from es.xml:
include
language name=es sound-path=$${base_dir}/sounds/es/ar/elianna
tts-engine=cepstral tts-voice=callie
Yes, I saw that after send that email.
This is the corrected one
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=directory
domain name=200.49.25.11
params
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain},transfer
hello,
i'm unable to load mod_xml_curl
thanks
Tim Panton a écrit :
On 17 Apr 2009, at 18:42, Meftah Tayeb wrote:
Hello,
please anyone here a implemented CURL for freeswitch using any Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL but
is not
hello,
the problem is: unable to load mod_xml_curl
thanks
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Brian, Michael,
Thanks for the help - I had read that but not fully comprehended it until you
spun it the way you did.
Here's what I ended up with - if there's optimization that could be done let me
know. Happy to update the wiki if this is a common request.
!-- Provide an internal ext
Hi, I am getting errors while installing FreeSwitch. I have enabled almost
all the packages (except: mod_snom, mod_ldap, mod_zap and few others). Do I
need to install dependencies or will FreeSwitch automatically find from the
internet? I have done the basic dependencies installed like gcc,
the issue was, its banned in the country i live, UAE as voip is considered
illegal here so i had bought it from india and i never got any file or
documentation from the company directly where i bought it from and i tried
like more than 200 different regional settings and i also set it up how UAEs
*Hi,
Michael Collins: I will explain in Detail
If the agent register 1000 in xlite.
The registered users can be viewed by api command api sofia status profile
default.
After that if i want to view the status of that extension 1000 whether he is
in calling are in ideal.
*
*There is any api
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