Before you go any further, could you let us know what you are trying to
accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do
you require some sort of authentication? Are the FS and Ast machines on the
same LAN?
It might help for you to pastebin the output from the FS CLI when
Dear,
I am just testing that how to connect FreeSWITCH with Asterisk. I don't
want any sort of authentication.
Yes, the FS and Asterisk are on the same LAN..
My intention is that When I call an extension from FS, the dial plan should
bridge a user in Asterisk..
Please give some suggestions
Don't forget you need to install libs/esl/perl/ESL.so and libs/esl/
perl/ESL.pm into your system perl library path.
/b
On Jul 28, 2009, at 1:53 AM, Michael Collins wrote:
> Make certain that you've built both libesl and the Perl mod. Change
> directory into /usr/src/freeswitch.trunk/libs/esl
Hello brian,
It was not exactly at the bottom but before
I tried to put it higher in the dialplan but it still doesn't work (with
the same error).
Thanks for your help.
Brian West a écrit :
> I have to guess that you put this at the bottom of the default.xml?
>
> /b
>
> On Jul 27, 2009, at 10
Hi everybody,
I intend to use Freeswitch with two Ekiga Softphones. SIP Instant
messaging works between the two softphones but SIP presence signaling
is not managed by the softphones. I try to use other softphones
(QuteCom and SIPCommunicator) and it is the same. I have the following
error
Well press F8 and increase the debug level.. then try again you'll
prob. see that its not finding it NOR matching it anywhere in your
dialplan.
/b
On Jul 28, 2009, at 4:32 AM, julien wrote:
> Hello brian,
> It was not exactly at the bottom but before
>
>
>
> I tried to put it higher in the
julien wrote:
> It was not exactly at the bottom but before
>
>
Why not put it in the default directory, from which it will be included by the
above line? If necessary, you could comment out any entries in default.xml
that might be matched first.
I've debugged this kind of problem before, and
You must turn on the option to manage presence in the sip profile.
Mike
On Jul 28, 2009, at 5:43 AM, Gregory Charles
wrote:
> Hi everybody,
>
>I intend to use Freeswitch with two Ekiga Softphones. SIP Instant
> messaging works between the two softphones but SIP presence signaling
> is no
Thanks for the tip Brian. It seems that the extension matches
successfully in the dialplan (PASS, instead of FAIL for all other
entries of the dialplan) :
Dialplan: sofia/internal/[EMAIL PROTECTED] parsing [default->pbxlyon]
continue=false
Dialplan: sofia/internal/[EMAIL PROTECTED] Regex (PASS)
The remote end said 404
/b
On Jul 28, 2009, at 10:00 AM, julien wrote:
> 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel
> sofia/external/300 entering state [terminated][404]
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FreeSWITCH-users mailing list
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Hello,
Please tell me, how can I execute originate new call and uuid_bridge in dial
plan.
I tried to make like thise:
result:
[ERR] switch_core_session.c:1239
switch_core_session_execute_application() Invalid Application originate
[ERR] switch_core_ses
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Hello,
today I finylly got a working Ekiga Softphone version which is able to
use high quality celt codec with FS :)
On my way to get it work with FS I found that Ekiga currently uses codec
code 95 in SDP while FS uses 114. Changing FS to 95 made it
What exactly are you trying to accomplish with this dialplan entry? That
will help us answer your question.
-MC
2009/7/28 Kozak Vladimir
> Hello,
>
> Please tell me, how can I execute originate new call and uuid_bridge in
> dial plan.
> I tried to make like thise:
> data="user/$${desti
On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote:
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Hello,
today I finylly got a working Ekiga Softphone version which is able to
use high quality celt codec with FS :)
On my way to get it work with FS I found that Ekiga currently uses
codec
cod
Здравствуйте, Kozak.
Try, for example
Вы писали 28 июля 2009 г., 18:16:45:
>
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in dial plan.
I tried to make like thise:
result:
[ERR] switch_core_session.c:1239 s
using 95 is wrong. That is not part of the dynamic range for
unassigned codecs. This needs to be fixed on their side.
MIke
On Jul 28, 2009, at 12:23 PM, Brian West wrote:
On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote:
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Hello,
today I finy
I totally missed this at first... but 95 wouldn't dynamically work
because its not 96-127
/b
On Jul 28, 2009, at 12:06 PM, Michael Jerris wrote:
> using 95 is wrong. That is not part of the dynamic range for
> unassigned codecs. This needs to be fixed on their side.
>
> MIke
_
Hello everyone,
I need to set a maximum call duration. What is the current
recommended way to implement this in FreeSWITCH? I'm looking for
something similar to AbsoluteTimeout() in Asterisk.
Thanks!
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.c
On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:
> Hello everyone,
>
> I need to set a maximum call duration. What is the current
> recommended way to implement this in FreeSWITCH? I'm looking for
> something similar to AbsoluteTimeout() in Asterisk
What needs to happen at the end of the timeout? In any case you can use the
sched_XXX APIs:
sched_api
sched_transfer
sched_hangup
You can get fancy or just hangup up on the call after X number of seconds...
:)
-MC
On Tue, Jul 28, 2009 at 10:28 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.
You can also schedule a playback then a hangup, what comes after the !
is the hangup cause.
sched_broadcast,Schedule a broadcast in the future,[+]
[aleg|bleg|both],mod_dptools
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.
also take a look at execute_on_answer if you want it to be scheduled
from answer instead of from that point in the dialplan.
Mike
On Jul 28, 2009, at 1:48 PM, Saeed Ahmad wrote:
On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner > wrote:
Hello everyone,
I need to set a maximum call du
A call to B.
A place to musikOnHold.
I wont play short musik rington B before make bridge A with B.
If use API comands, it's work:
//1. A to musikOnHold.
sendmsg
call-command: execute
execute-app-name: execut
All awesome replies.
Thanks again!
On Tue, Jul 28, 2009 at 1:59 PM, Michael Jerris wrote:
> also take a look at execute_on_answer if you want it to be scheduled from
> answer instead of from that point in the dialplan.
> Mike
> On Jul 28, 2009, at 1:48 PM, Saeed Ahmad wrote:
>
>
>
--
Kristian
Hello,
If I wanted a bridged call to a gateway to use inband DTMF for
incoming recognition and outgoing generation I'm unclear on what to do
because the wiki clearly states[1] not to use the "start_dtmf" and
"start_dtmf_generate" together for cause of loops.
Wouldn't it be technically possi
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Hi,
I've just tried new mod_managed under Win32 and I get a weird behavior.
I try the example below:
public class DemoScript : IApiPlugin
{
public void Execute(ApiContext context)
{
context.Stream.Write(string.Format("DemoScripts exec
Hi,
I'm doing some tests between 2 FS to understand how the tone_detect application
works.
I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only
get the first activated tone.
If I have all tones activated - - - > I detect only the first tone of my wav
file.
If I have
Hi, [Thanks for the advice Anthony]
I tried "send_silence_when_idle=true " and restarted, but did not notice
any change/improvement.
But I had limited time to test, so will need to test more thoroughly
with this CPE.
An additional test was to configure the following media path scenarios:
Hello Lukasz,
Thanks for testing mod_managed. I apologize for the problems you've
encountered, and I'll try to sort them out for you.
A few things first:
- Scripting support: This is made to allow "true" scripts, as invoked
as an EXE - similar to the Lua and spidermonkey suppor
Have a look at the sip traffic.
I believe in the default configuration that FreeSWITCH will use negotiated
CNG (payload 13) if the other end supplies it.
The description you got from your vendor is entirely accurate but they are
supposed to handle this situation.
When we stop sending RTP for a dur
Hello, I'm looking for a way to bridge 2 external PBX devices without keeping
the FS gateway occupied during the conversation.
My configuration:
FS is registered as a sip-endpoint ("G - gateway") in a regular PBX, gateway
configured on FS side, inbound and outbound working fine. "G" is limited to
Brian West wrote:
> I totally missed this at first... but 95 wouldn't dynamically work
> because its not 96-127
>
> /b
>
The way to CELT plugin registers should be ok, the code is based on the Speex
plugin.
Opal tries to assign a number in the dynamic range first and if nothing is free
in t
Hi Brian,
Sorry responding late. I still cannot get this work, can you take a look?
http://pastebin.freeswitch.org/9877
Everything works fine on Linux but not on my MAC. I have the default ruby
framework and port install on /opt/local/bin/ruby, however, even I changed
the Makefile to use the def
Just to mention, there is also a Ruby library here for FreeSWITCH, similar
to ESL, it might interest you.
http://code.rubyists.com/projects/fs
http://github.com/bougyman/freeswitcher/tree/master
http://blog.rubyists.com/2009/05/19/ruby-freeswitch-love
On Wed, Jul 29, 2009 at 12:45 AM, Seven Du w
Dear all,
For implementing a IVR I planned to use event outbound socket.For that I am
in the process of analyzing the /libs/esl/perl/server2.pl code.In that they
created a object for ESL::ESLconnection package then they called some of the
methods like getInfo,sendRecv ...etc using that object.
S
Hi Michael ,
thanks a lot for support on this.
> As to the main problem of your DLL not working, can you send me the full
> source code, or all the logging output from loading it? Try "managedreload
> my.dll" to reload the DLL and see how it is registering them. It should
> output something li
Hi Łukasz,
Would you please send me the DLL offlist and I'll figure it out?
The new session you create is the b-leg. The parameter it takes in
originate is the a-leg. So you'd do:
var session = new ManagedSession();
session.Originate(context.Session, "sofia/default/1000",10);
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