Hi Guys,
I'm placing calls ok by using the event socket. However, I need to
modify the To: Sip header prior to the call going out for routing
purposes. Is it possible to do this in the Originate action?
If not, can someone explain if it's possible to trigger part of the dial
plan
Hi Nik,
How do you need to modify it?
Cheers --
Dave
Hi Guys,
I’m placing calls ok by using the event socket. However, I need to
modify the To: Sip header prior to the call going out for routing
purposes. Is it possible to do this in the Originate action?
If not, can someone explain
It seems I know the same you know, it was on the works then not, then
back in the works. However I don't know the status on that. If you
have contact with Doug he is a better person to ask to regarding new
products coming out.
Moy
On Fri, Feb 6, 2009 at 9:16 AM, Shelby Ramsey sicfsl...@gmail.com
Sent from my iPhone
On Feb 7, 2009, at 9:03 PM, Paul D. pa...@versafon.com wrote:
Followed Wiki to install and configure mod_cepstral. The problem is FS
always defaults to one voice, which I installed first, and ignores
others.
I did define SWIFT_HOME and added swift lib path to
Hi Guys,
I'm having some issues passing an argument to an lua script.
Basically in an originate command I pass the name of a .wav file
If I hard code the file session:streamFile(myfile.wav]);
It works,
However, using this:
session:streamFile(argv[1]);
causes this error
Hi Guys
I want to access Mysql 5 from lua. The wiki is not too clear on this.
Do I need to install lua and lua mysql?
Regards
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The libs are there with correct symlinks, see below. I tested both
voices directly via swift command, works fine.
Any other ideas?
It's Cepstral 5.1, FS 1.0.2.
ls -la /opt/swift/lib
total 4352
drwxr-xr-x 2 root root 4096 Feb 7 22:59 .
drwxr-xr-x 10 root root 4096 Feb 7 12:29 ..
Print out the variable to make sure it is what you expect:
io.write(argv is .. argv[1] .. \n;
Also, if you don't give the sound file an absolute path name then it
will automatically use the sound dir path.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:31 PM, Nik Middleton
Nik,
I see your point about the wiki entry regarding luasql. If someone
could clarify then I will be happy to help get the wiki documentation
updated appropriately.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:41 PM, Nik Middleton nik.middle...@noblesolutions.co.uk
wrote:
Hi Guys
Done that, still doesn't work. My guess is related
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael S Collins
Sent: 08 February 2009 23:21
To:
If you're wanting to modify the invite domain you can do that via the
sip_invite_domain variable either export it before the bridge or place
it inside {} on the originate line... ie
{sip_invite_domain=example.com}sofia/blah/blah
/b
On Feb 8, 2009, at 11:09 AM, David Knell wrote:
Hi Nik,
Looks like you put a , instead of a space when calling the script.
/b
On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote:
cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav
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Hi,
Right now, I am working on getting freeswitch configured for our call-center
with more than 1000 agents. There are several areas where we need the
dialplan to be configurable based on some user detail in the database.
Therefore, the dialplan needs to be some-what dynamic based-on inputs from
If you want an extremely high performance you write your own dialplan
module... Its not that hard...
Or option 1 is the more high performance way to fo...
Curl with a serverlet will scale to a point but I doubt it will get to where
you need in the long run
Static and do what you need, but how
Doug,
Ken is right on this one. I know there are some guys on the list (like Ken)
that could help you write a module. It's probably the best way to go (if
you're going to have all agents running off of one or two boxes).
If you're going to spread the agents / calls around on multiple boxes or
Also depending on what your Timeframe is like there is a distributed queue
mechanism with skills based routing on the way...
From: Shelby Ramsey sicfsl...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Sun, 8 Feb 2009 22:20:22 -0600
To: freeswitch-users@lists.freeswitch.org
On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote:
Also depending on what your Timeframe is like there is a distributed queue
mechanism with skills based routing on the way...
It even managed to route 2 calls in a row this week ;) Still a ways off
from anything production grade tho.
On Mon, Feb 9, 2009 at 10:16 AM, Andrew Thompson and...@hijacked.us wrote:
On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote:
Also depending on what your Timeframe is like there is a distributed
queue
mechanism with skills based routing on the way...
It even managed to route 2
Hello everyone,
Now that I've got FreeSWITCH compiling under AstLinux I'm starting
to look at ways to optimize FreeSWITCH.
First things first: minimize transcoding. I hate transcoding.
I modified a concept I came up with back in the day for Asterisk.
I've created a script to convert WAV
Hello,
in my reallife setup of FS all internal extensions use G.722 as
preferred codec. Unfortunately when there is an outgoing TDM call I
found that FS starts transcoding instead of forcing G.711 for A leg. So
is there a way to force the A codec?
regards
helmut
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