Awesome work, Kristian!
And very much needed for the Freeswitch platform (to me, at least).
A suggestion: if the FS team doesn't mind (after getting over the naming
issue), it would be a good idea to put Kristian's latest blog entry on the
FS Wiki.
-Original Message-
From:
Hi Anthony,
I tried your second option, but how does it work with xml-curl then ?
As far as I understand it, this doesn't work by doing a user-directory
xml lookup at INVITE time, or does it ?
Or does it want to generate an ACL at FS startup and filling up all
the allow-nodes by polling
if the inbound calls are coming from a registration to a provider you will
have to set a context param in the gateway itself.
All inbound calls from a gateway registration are now associated with the
gateway they were registered with and inherit
the context from there.
Maybe i'll change the
Hi all,
I'm able to receive a fax with mod_fax, but I still don't understand how
to send fax.
I don't understand how to send the fax through a specific profile/IP. Is
mod_fax limited to an openzap interface??
when I dial to the extension with tx_fax, I get a tone then hangup, but
what I'd
done,
r12138 should give you the correct behavior
On Wed, Feb 18, 2009 at 8:09 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
if the inbound calls are coming from a registration to a provider you will
have to set a context param in the gateway itself.
All inbound calls from a
yes that is correct.
On Wed, Feb 18, 2009 at 8:03 AM, Leon de Rooij l...@scarlet-internet.nlwrote:
Hi Anthony,
I tried your second option, but how does it work with xml-curl then ? As
far as I understand it, this doesn't work by doing a user-directory xml
lookup at INVITE time, or does it ?
Anthony Minessale napsal(a):
done,
r12138 should give you the correct behavior
Thank you very much, Anthony! Incredible speed :-)
Best regards,
kokoska.rokoska
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OSX can be loaded on any new Intel machines..
-E
http://TwiTR.Me
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And one more thing...
As soon as it recognizez TAKEOUT, freeswitch crashes...
2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log()
Heard []
2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log()
Hit score 98/40/70
2009-02-18 19:03:51 [INFO]
Please update to SVN Trunk and try again... what are the specs on your
machine?
I have been testing PocketSphinx the past couple of days on linux
again and its fine.
/b
On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote:
Hi,
I have downloaded and build the Freeswitch from
Please go get an SVN client for windows... svn update vs downloading
the tarball every day will save bandwidth. ;)
/b
On Feb 18, 2009, at 9:49 AM, Moiz Chinoy wrote:
System specs:
- Intel Core 2 Duo
- 2.00 GHZ CPU
- 1 Gb Ram
I will download the latest from here
On Wed, Feb 18, 2009 at 3:21 PM, EdPimentl edpime...@gmail.com wrote:
OSX can be loaded on any new Intel machines..
-E
That's nice! How I can do it, I mean, in an easy way that will let me
develop on it? It's just like installing a distro, or involves black
magic?
-gm
On Wed, Feb 18, 2009 at
Hi Rod, i just play with rx_fax and work for me. I didn't work with tx_fax
but i understand, that you need a .tiff file to send passthrough the rx_fax.
Maybe that can help you
regards
javar
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bandwidth.com has a service called phonebooth which is developed upon
freeswitch.
On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso r...@etellicom.com wrote:
Hello FreeSWITCHERS,
My company is currently creating a suite of applications which uses
FreeSWITCH as the back-end for an IP-PBX
Hi Raul,
In my company (http://www.globant.com) we're using FreeSWITCH for high
quality conference services, integrated with OpenSIPS
(http://www.opensips.org) and Asterisk. Its performance is pretty
good.
Pablo
On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang red.rain.se...@gmail.com wrote:
Nik,
What are you building? I'm wondering if this is the correct approach
for your application. You might be better off using the even socket
and controlling your calls from a central point.
-MC
On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
On Wed, Feb 18, 2009 at 7:55 AM, Brian West br...@freeswitch.org wrote:
Please go get an SVN client for windows... svn update vs downloading the
tarball every day will save bandwidth. ;)
/b
Use this for windows:
http://tortoisesvn.tigris.org/
-MC
In that case you would need a sip proxy in place to rewrite the packets for
the nat issue.
There's nothing else we can really do. We have a way to do what you want
but you are using it under
circumstances we can't control.
On Wed, Feb 18, 2009 at 1:46 PM, Justin Miller
I'm trying to build an emergency broadcasting solution.
So I place a call, and have ivr in the lua script. But I also want to
give them the option of speaking to someone.
If they hit the option to speak to someone, while I can fire an event to
originate a call, I'm not sure how I could bridge
On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
I'm trying to build an emergency broadcasting solution.
So I place a call, and have ivr in the lua script. But I also want to
give them the option of speaking to someone.
If they hit the option to
Hi Michael,
Yes that's exactly what it boils down to, an outbound ivr.
Everything is working perfectly, except the bridge to another number.
Because of the nature of the beast the bridge needs to dial an external
number (ie sofia/gateway/Mygateway/num) What I'm getting is:
attempt to perform
Everything is working perfectly, except the bridge to another number.
Because of the nature of the beast the bridge needs to dial an external
number (ie sofia/gateway/Mygateway/num) What I'm getting is:
attempt to perform arithmetic on global 'sofia' (a nil value)
Can you pastebin your Lua
http://www.tech-recipes.com/rx/964/install_osx_tiger_on_intel_usb_drives_windows/
-E
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Sorted now thanks, it needed to be in the format
session:execute(bridge, {params}sofia/gateway/Mygateway/number);
key change was ''
Now I've converted my js script to lua going to run some tests tomorrow.
I sincerely hope it'll handle more than the 10 calls js would break at.
Here's my
Learn C and write it all in C.
/b
On Feb 18, 2009, at 3:56 PM, Nik Middleton wrote:
Astererisk happily does around 200 calls, I'm hoping FS will do better
or I've just been wasting my time. Is there a more efficient way of
doing this?
___
You want to make it even more efficient?
when they press 1,
session:execute(transfer, number);
Then, put an extension in your dialplan to match number and do the bridge.
Then you can exit the script and only run the script when you need it.
Your problem with js was the same issue, you should
i replied to your last private message and it was returned as
undeliverable. overzealous spam server? Can you add my account to your
whitelist?
On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
You want to make it even more efficient?
when they press 1,
Done
Seems it had a spam score of 2 for some reason
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 18 February 2009 23:39
To:
Hi All.
Have a fresh server and going to install FS on it. Went to the download
page (http://wiki.freeswitch.org/wiki/Installation_Guide) and tried to
download the Phoenix build, which is supposed to be found at
http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but that file is nowhere
to be
Giovannia,
great work on mod_skypiax. I've been testing it under Windows and it sounds
great including PSTN calls. I plan to include it as part of the Windows MSI
build.
One question I have, is ringback suppose to work with mod_skypiax? Whenever
I dial a number I get a few seconds of dead air
Looks like someone jumped the gun... just get SVN trunk... we are in
the process of release right now.
/b
On Feb 18, 2009, at 8:00 PM, Philip Patterson wrote:
Hi All.
Have a fresh server and going to install FS on it. Went to the
download page
Thanks guys, this is very useful information.
Anyone else willing to share your experience ?
Regards,
Raul
On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote:
Hi Raul,
In my company (http://www.globant.com) we're using FreeSWITCH for high
quality conference services, integrated
On Wed, Feb 18, 2009 at 6:57 PM, Brian West br...@freeswitch.org wrote:
Thats one I think Anthm will need to chime in on... maybe skypiax
isn't sending the right indications to cause the core to trigger the
ringback.
/b
Out of curiosity, you might try this trick:
action application=set
That did it!
I had to add both lines below in order for it to work:
action application=set data=instant_ringback=true/
action application=set data=ringback=${us-ring}/
Now, suppose I call a number that's busy...do I hear a ringback followed by
a busy signal?
On Wed, Feb 18, 2009 at
Carlos,
Maybe the solution Michael is suggesting could work.
For sure you are not missing anything' Brian is right: rimgback and
early media are to be added to skypiax. They're on the TODO section of
the wiki :-)
I'll be all day at a customer's premise, I'll add it this evening,
late afternoon for
go try now! ;)
/b
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It has to be in trunk to be in the MSI... I don't want to cause
confusion ... Now that 1.0.3 is tagged we can put it in trunk?
/b
On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote:
Would be *very* nice to have skypiax in MSI, thank you!
___
Thanks for your help
I have downloaded the latest build and tried...
Often in the log I see cryptic characters in the XML part returned by
ASR. Is it silence or nose??
If yes is there any way we can control it?
Prompt playback problem is still there...
So far I am only able to get TAKEOUT
Yes, I'd like it in trunk.
There are still some rough edges, but I'll iron out in the trunk.
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039
No clue, I haven't ever seen that behavior on linux. Maybe you can
try to narrow it down and report it on jira.. chances are its a bug in
the pocketsphinx libs.
/b
On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote:
Often in the log I see cryptic characters in the XML part returned by
ASR.
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