Hi,
you could build an embedded filesystem using one of these great tools:
- buildroot crosstools-ng
- ptxdist
- moblin2
- clfs
The one i prefer is buildroot because is simply, flexible and fast to
build a root filesystem: to setup a minimal system, once you builded the
toolchain using
We currently use Patton gateways SN4116 for attaching fax and modem
equipment to our Freeswitch system. Freeswitch is in bypass-media-mode,
so media flow goes the following way:
Modem/Fax = Patton_SN4116 = Patton_SN46XX =PSTN/ISDN
However modem connections are not very reliable. We exchanged the
On 12/14/2009 08:05 PM, Peter P GMX wrote:
We currently use Patton gateways SN4116 for attaching fax and modem
equipment to our Freeswitch system. Freeswitch is in bypass-media-mode,
so media flow goes the following way:
Modem/Fax = Patton_SN4116 = Patton_SN46XX =PSTN/ISDN
However modem
Hi,
What are the solutions to support the G729/G723 codec within FreeSwitch ?
Thanks
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Hello
I'm a bit confused at the difference between those two concepts. Contexts
are created in the /dialplan, and are refered to by items in /SIP_profiles
and extensions in /directory.
What purpose do contexts and profiles play?
Thank you.
--
View this message in context:
Software G729 will be available by the end of the month.
As for, G723 we are not currently working on it.
On Mon, Dec 14, 2009 at 6:45 AM, Oscav os...@hotmail.fr wrote:
Hi,
What are the solutions to support the G729/G723 codec within FreeSwitch ?
Thanks
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View this message in
Profile is a collection of preferences uses by conferences etc.
In the case of SIP a profile is also the name for the resulting SIP UA
created by a particular profile.
Context is a narrowed down view of something, in the case of the dialplan a
context is a set of extensions. It's like having a
a dial plan is a another level of indirection on top of contexts it denotes
a specific module which implements the entire universe of dialing with room
for as many contexts as you have room for. There is an XML dialplan, an
ENUM dialplan etc. You can write your own dialplan and send calls to it.
On 12/14/2009 11:19 PM, Anthony Minessale wrote:
Software G729 will be available by the end of the month.
As for, G723 we are not currently working on it.
There is a legit option for G.723.1 - the Digium TC400B card. Its
supported by Freeswitch, thanks to Moises.
On Mon, Dec 14, 2009 at 6:45
Both the INVITE and the 200 OK have an a=crypto line. You need to know
that only the rtp payload is encrypted, it is normal that you see the
headers as-is.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 14-Dec-09, at
Thanks Anthony for the tip.
Would you say this is a correct representation of things?
Contexts are a set of extensions in conf/dialplan/ (eg. default, public,
etc.)
Extensions are configured through files in conf/directory/. Each extension
maps to a context (variable name=user_context
Profile is a collection of preferences uses by conferences etc.
In the case of SIP a profile is also the name for the resulting SIP UA
created by a particular profile.
Context is a narrowed down view of something, in the case of the dialplan a
context is a set of extensions. It's like having a
On Mon, Dec 14, 2009 at 9:12 AM, Fred-145 codecompl...@free.fr wrote:
Thanks Anthony for the tip.
Would you say this is a correct representation of things?
Contexts are a set of extensions in conf/dialplan/ (eg. default, public,
etc.)
Extensions are configured through files in
HI
I tried to build the svn last Friday and it didn't make the sphinx dll, so I
thought I would wait for the Monday build (web site says update every Monday).
Is there going to be a windows build today, and or is the sphinx dll
build problem fixed if I build it myself?
Thanks!!
I do pre releases and it'll be up shortly had to fix a couple of
bugs. I don't do binary releases for windows you'll have to do that
yourself or wait.
/b
On Dec 14, 2009, at 12:45 PM, Kendall Stauffer wrote:
HI
I tried to build the svn last Friday and it didn’t make the
sphinx dll,
Hey everyone,
I am researching billing solutions for Freeswitch and want to consolidate
the information with what others have found, then add it to the Wiki.
There are seems to be a number of billing solutions by commercial providers,
claiming they can integrate with Freeswitch, but nothing
Thanks for your answer,
I want to resolver the caller id to dialplan level, I know variables
fifo_caller_consumer_import and fifo_consumer_caller_import but isn't clear for
me how use it.
Diego Toro
http://lacarretade.blogspot.com/
--- On Sun, 12/13/09, Seven Du dujinf...@gmail.com wrote:
Please post back to the list if you have problems with the windows build!
Everything is working as far as I know.
If you have an existing build you should delete the following directories
and let the scripts download it again.
libs\pocketsphinx-0.5.99 - delete
libs\sphinxbase-0.4.99 - delete
FYI,
The latest pre-release is now available. Usual information is available
here: http://www.freeswitch.org/node/222
Please update as soon as you can. (SVN trunk users do the make current
thing please.) We need your testing and feedback please!
Many thanks for continuing to support FreeSWITCH.
I'm playing with demo IVR from FreeSwitch distribution and have a
problem with language settings. I would like to use Russian as a
default language for voice messages so I set in vars.xml
X-PRE-PROCESS cmd=set data=default_language=ru/
and installed Russian sound files. It works almost
Hello everyone,
I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like
to connect to the event socket and run some api commands and return
them (as opposed to checking SIP, for example). I haven't found
anything and I've started to write one in perl using ESL. I'm sure
whatever I
Also Pre9 is up now.
/b
On Dec 14, 2009, at 1:25 PM, Jeff Lenk wrote:
Please post back to the list if you have problems with the windows
build!
Everything is working as far as I know.
If you have an existing build you should delete the following
directories
and let the scripts
On Mon, Dec 14, 2009 at 11:49 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like
to connect to the event socket and run some api commands and return
them (as opposed to checking SIP, for example).
On Sun, Dec 13, 2009 at 5:27 PM, David V. Fansler
dfans...@dv-fansler.comwrote:
Final Count was just over 900 files. At the moment I am putting them in
a logical order – as best I can tell with my limited experience – combining
chapters and providing links from the table of contents to
On Sun, Dec 13, 2009 at 9:10 PM, Thangappan.M thangappan...@gmail.comwrote:
I've seen the source code of mod_rss and mod_voicemail. I was not able to
get it.
What are the steps do I need to follow for implementing IVR using IVR
library?
Is there any documentations available for knowing
On Fri, Dec 4, 2009 at 7:58 AM, Neil Patel ne...@cs.stanford.edu wrote:
Hi All,
I haven't found a substantial example of IVR applications implemented in
lua. Can anyone suggest where to look? My issue has to do with appropriate
coding style.
I am implementing a voice message board
On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely dmitry.b...@gmail.com wrote:
I'm playing with demo IVR from FreeSwitch distribution and have a
problem with language settings. I would like to use Russian as a
default language for voice messages so I set in vars.xml
X-PRE-PROCESS cmd=set
you need to get the fd number of stdin however you do it in sdp and pass it
as the constructor to the esl obj
On Sun, Dec 13, 2009 at 8:41 PM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
Now i use php for ESL outbound. i get variable from stdin and
process. (i use xinetd for
Hi Anthony,
What kind of software?? Is there any related licensing cost? Will it be also
available for windows ??
Regards,
Oscav
Anthony Minessale-2 wrote:
Software G729 will be available by the end of the month.
As for, G723 we are not currently working on it.
On Mon, Dec 14, 2009
Will do - once I have the toc compiled, I would like someone more knowledgable on FreeSWITCH to see if the order makes sense.
Thanks,
David
-Original Message- From: Michael Collins Sent: Dec 14, 2009 12:20 PM To: freeswitch-users@lists.freeswitch.org Subject: Re:
Hmm, I emailed the zfoneproject folks about an hour ago asking about a
release date for zfone3 and was surprised about a half hour later with a call
from PRZ himself.
Here's what I got from the call
1) the currently released version of zfone already has support for secure pbx
enrollment
2)
if you don't have ZRTP compiled in as per the wiki it won't work...
their are a few changes coming to this code soon.
/b
On Dec 14, 2009, at 8:01 PM, Harondel J. Sibble wrote:
Hmm, I emailed the zfoneproject folks about an hour ago asking about a
release date for zfone3 and was surprised
Hi All,
I am trying to setup a conference room with pin number authentication. I
could not find any wiki documents. If some one help me that would be
helpful.
Thank you in advance.
Thanks
Lloyd
___
FreeSWITCH-users mailing list
http://wiki.freeswitch.org/wiki/Mod_conference
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 14-Dec-09, at 10:06 PM, Aloysius Thevarajah Lloyd wrote:
Hi All,
I am trying to setup a conference room with pin number
I do have it compiled in as per the wiki ;-)
That's why I am continually scratching my head on why it's not working.
I did have to use the scrooge codec setup otherwise the enrollment messages
would play at about 1/10 normal speed. :-(
On 14 Dec 2009 at 20:50, Brian West wrote:
if you don't
Great job. You would hear lots of thanks, if you could hear the entire
world.! Thanks.
Regards,
Nagalenoj H.
On Mon, Dec 14, 2009 at 6:57 AM, David V. Fansler
dfans...@dv-fansler.comwrote:
Final Count was just over 900 files. At the moment I am putting them in
a logical order – as best I
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