Jed,
here are the list of things you can do:
1. sip_profiles/default.xml -> change context to default and set auth-
calls=false
2. Then you can use ${network_addr} in your conditions or the ${acl()}
function an example is in the default.xml dialplan.
/b
On Apr 29, 2008, at 4:44 PM, J
This feel's like a very stupid question, but i've scoured for hours through
the documents, and samples I can find without finding an answer. I'm
assuming I'm missing something very obvious.
I'm just trying to have freeswitch accept a call from a static IP address,
then forward that call to a provid
> I scheduled it for 2 hours but for most of them they'll be 1 hour..
> but that should give us some spare time.
>
> Can you pick 10-20 API's and Apps that we can go over and verify all
> the docs on the wiki...
I'll throw in a few...
It seems there've been some recent threads about how to handl
I scheduled it for 2 hours but for most of them they'll be 1 hour..
but that should give us some spare time.
Can you pick 10-20 API's and Apps that we can go over and verify all
the docs on the wiki...
/b
On Apr 29, 2008, at 1:43 PM, Michael Collins wrote:
> I'd like to try one tomorrow. I
I'd like to try one tomorrow. I don't know if I can stay on for two
solid hours but I will give it my best shot.
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-dev-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Tuesday, April 29, 2008 11:22 AM
> To: freeswitc
Lets try every wednesday at 1pm CST. Do you want to try one tomorrow?
/b
BEGIN:VCALENDAR
X-WR-TIMEZONE:US/Central
PRODID:-//Apple Inc.//iCal 3.0//EN
CALSCALE:GREGORIAN
X-WR-CALNAME:FreeSWITCH Conference Call
VERSION:2.0
X-WR-RELCALID:E8A3289D-A6B2-4FC7-934E-FA5327830156
METHOD:PUBLISH
BEGIN:VTIM
Cavalera -
You'll find it much easier to use the XML-RPC for this sort of
thing, I think. I'm sure there is a php library for XML-RPC and then
its just one or two lines of code.
On Apr 29, 2008, at 10:59 PM, Cavalera Claudio Luigi wrote:
> I like the idea of contribute actively to fs devel
I like the idea of contribute actively to fs development, however at the
moment I can contribute only report my experience as user. In fact I am
a PHP newbie anyway, I post here my script in case it can be useful to
anyone.
It still needs a lot of work, but it's a rough start.
Caller is .
Callee i
Yes,
I'm trying to do 1) "Web callback" at the moment :-)
BRs,
Claudio
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
EdPimentl
Sent: Tuesday, April 29, 2008 3:41 PM
To: freeswitch-users@lists.freeswitch.org
Subjec
Claudio -
I can give you some pre-release Ruby on Rails code that, if you
are developing your web app in Rails will do the trick in a few
lines. Specifically
class CallController < ApplicationController
def create
VoiceChannel.create(:destination => "sip/default/#{params
[:user1]}",
This is really the basis of CallBack
Here is what one can do with CallBack
· You tell the service what telephone number you are calling from.
· You tell the service what telephone number you want to call.
· The service dials both (you first) and connect the two.
Remember, since you are being ca
Larry,
I would recommend you get on IRC #freeswitch @ irc.freenode.net and
the guys can assist you in getting fast tracked to a working system
then maybe we can work out why the documentation wasn't able to help
you and correct it.
Thanks,
Brian West
On Apr 28,
That sounds like we may not be injecting silence into the file during voice
inactivity.
We will have a look.
On Tue, Apr 29, 2008 at 7:33 AM, Jonathan Palley <[EMAIL PROTECTED]>
wrote:
> Hello all -
> I'm using uuid_record to record calls. We are encountering a
> problem where the voice for ei
The correct answer is "take your pick"
Every mechanism you described is a thin layer on top of the same core code
so they all work the same way.
The FSAPI interface is designed to be connected to any lightweight control
protocol.
With all the time you saved not having to wrestle with a solution t
Hi all -
I installed Freeswitch a few weeks ago and was never quite able to connect
to it via Gizmo or IPKall. I can't seem to configure the directory
correctly and get my Gizmo phone to register; my IPKall phone doesn't seem
to get anywhere either. I'm a little fuzzy on the exact error messages
Hello all -
I'm using uuid_record to record calls. We are encountering a
problem where the voice for either channels starts overlapping and
not being in sync as it is in the actual call. I'm guessing perhaps
I'm missing a settings to make sure the calls are mixed in realtime
and record
Hello,
I'm trying to realize a demo for this use case:
1) a user inserts two sip users (e.g. 1001 and 1002) into a web page
2) fs calls the users (already registered to fs) and bridge them
I've succeeded to achieve 2) by a mix of commands such as:
originate sofia/default/1001%131.132.133.134 & pa
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