Dear All,
I have tried to connect the FreeSWITCH with Asterisk
I have followed steps which is provided in the following link,
http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk
I have tried to call 2000 from FreeSWITCH, but I have received the
following message in Asterisk
Dear all,
In the previous post, I got the information that using event outbound
socket we can implement the IVR and also see the example in
libs/esl/perl/server2.pl.
I have seen it and understood the flow of the script.But when I was
running that script it tells the following error.
Make certain that you've built both libesl and the Perl mod. Change
directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your path is
to libs/esl) and do these commands:
make
make perlmod
Then give it another shot.
-MC
On Mon, Jul 27, 2009 at 11:16 PM, Thangappan.M
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
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
X-PRE-PROCESS cmd=include data=default/*.xml/
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
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
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
X-PRE-PROCESS cmd=include
julien jgonza...@sqli.com wrote:
It was not exactly at the bottom but before
X-PRE-PROCESS cmd=include data=default/*.xml/
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
You must turn on the option to manage presence in the sip profile.
Mike
On Jul 28, 2009, at 5:43 AM, Gregory Charles
gregory.char...@sogeti.com wrote:
Hi everybody,
I intend to use Freeswitch with two Ekiga Softphones. SIP Instant
messaging works between the two softphones but SIP
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]
___
FreeSWITCH-users mailing list
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in dial
plan.
I tried to make like thise:
action application=originate data=user/$${destination_end_point}
playback(${hold_music})/
action application=originate data=user/$${destination_end_point},
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
What exactly are you trying to accomplish with this dialplan entry? That
will help us answer your question.
-MC
2009/7/28 Kozak Vladimir vko...@abisoft.spb.ru
Hello,
Please tell me, how can I execute originate new call and uuid_bridge in
dial plan.
I tried to make like thise:
On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
Здравствуйте, Kozak.
Try, for example
action application="bridge" data=""/
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:
action application="originate" data=""/
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:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today I
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
action application=sched_hangup data=+600/
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
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
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,[+]time path
[aleg|bleg|both],mod_dptools
action application=sched_broadcast data=+600 playback!
normal_clearing::/path/to/file /
Mathieu Rene
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:
action application=sched_hangup data=+600/
On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner
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:
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 possible
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
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
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
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
2
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 that
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
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
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