I'm trying to get output to my CLI in Windows XP when running this script by
dialing 8337 but the output "Hello World!" doesn't show up. The rest works fine.
What's wrong with console_log("Hello World!\n") ? Where has the output gone?
console_log("Hello World!\n");
var languageCode = "en";
var
On Wed, Oct 29, 2008 at 7:20 AM, Peter P GMX <[EMAIL PROTECTED]> wrote:
> I did a svn log:
>
> /usr/src/freeswitch/libs/apr# svn log
>
> r9605 | mikej | 2008-09-20 02:05:00 +0200 (Sa, 20 Sep 2008) | 1 line
>
> hack for now un
I did a svn log:
/usr/src/freeswitch/libs/apr# svn log
r9605 | mikej | 2008-09-20 02:05:00 +0200 (Sa, 20 Sep 2008) | 1 line
hack for now until we ditch apr dso code completely
There is something that a patch to apr we did broke and we are working
on it still.
Do svn log on libs/apr and revert the last patch for a temp fix
On 10/28/08, Michael Collins <[EMAIL PROTECTED]> wrote:
> It would seem you are ahead of me... sorry I couldn't be of further
> assistance.
> -MC
>
>>
I am in the process of making my FreeSWITCH installation highly
available and I'm running into a couple of snags that was hoping that
someone may have some insight on.
First, the setup as it is now.
There are two installations of FS on two different servers, lets call
them fs1 and fs2. They
It would seem you are ahead of me... sorry I couldn't be of further
assistance.
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Peter P GMX
> Sent: Tuesday, October 28, 2008 4:36 PM
> To: freeswitch-users@lists.freeswitch.org
> Subjec
I have only German sound files at present. So I symlinked them also to
libceplang_en.so*
libceplex_en.so*
and did an ldconfig again.
Here is a list of libs in /opt/swift/lib
lrwxrwxrwx 1 root root 20 2008-07-17 18:10 libceplang_de.so ->
libceplang_de.so.5.1
lrwxrwxrwx 1 root root 20 2008-07-17
Bummer.
I had some issues with the 5.0 version but that was because of file
naming issues. I finally created symlinks and got it working. I've not
tried 5.1 yet. When I get a minute I will and I'll see if I get the same
error or not.
For kicks, can you try a different language? I'm just curious t
Hello Michael,
No, I startet with a 5.1 installation.
Cepstral works on the command line
opt/swift/bin/swift -o hello.wav 'Hallo Peter'
And the voice is registered:
[EMAIL PROTECTED]:/opt/swift/bin# ./swift --voices
Swift command-line synthesis program
Version 5.1.0 of July 2008
Copyright (c) 2
> Hello,
>
> I receive the following message during CS_INIT
>
> *Failed to load library libceplang_de.so due to:
> /opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int*
>
Hmm... that's odd. Did you have an older version of Cepstral installed
prior to going to 5.1?
-MC
___
Hello,
I receive the following message during CS_INIT
*Failed to load library libceplang_de.so due to:
/opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int*
Later however, FS at least tries to speak:
2008-10-28 23:40:05 [NOTICE] mod_dptools.c:605 answer_function() Channel
[sofia/intern
Woof!
I used to get lots of variable_* lines when using socket_outbound. They
have disappeared. Is there something I need to configure to get them back?
--Woof!
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On Oct 28, 2008, at 3:26 PM, Ryan McDougall wrote:
> Apologies if this has been answered somewhere already, but does
> freeswitch expose an API that would make it appropriate as a SIP
> proxy?
Apologies if this has been answered somewhere already, but does
freeswitch expose an API that would make it appropriate as a SIP
proxy?
Any advice you could share would be greatly appreciated.
Cheers,
___
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Freeswitch-users@
Thanks, that is the solution I am working on now... and the one I wanted to
avoid. At the moment, I am doing this with kamailio and asterisk. As far
as I get deep knowledge on FS I will replace Asterisk.
Regards.
2008/10/22 kokoska rokoska <[EMAIL PROTECTED]>
>
>
>
> Arturo Díaz Almagro naps
Hi Folks,
I need some additional help with this issue. I already had some from Brian i'm
but still not able to move forward.
I want a non-registered device to be able to call extension 56900 in my
Freeswitch in such a way that i can manage the call using the socket interface.
I believe the is
I got FreeSwitch running today and connected a softphone to it. Yippee!
But ran into some snags trying out Javascript within a dialplan.
My FreeSwitch is running on Windows XP and from the same machine I use
an X-Lite softphone to call and play around while learning how
Javascript and
Hi Anthony,
Yes, it works now!!!
I added an additional 'while (session.ready())' loop in the consumer
script. I also had to add 'setAutoHangup(false)' to make it work.
For archive purposes, here is the latest version of my javascript function:
// conf_name: the name of the conference, for exa
can you add more debug to the following places (with your patch disabled)
ozmod_ss7_boost.c:245
zap_log(ZAP_LOG_CRIT, "setting init state to progress_media\n");
zap_io.c:999
zap_log(ZAP_LOG_CRIT, "init_state=%d\n", zchan->init_state);
mod_openzap.c:368
zap_log(ZAP_LOG_CRIT, "calling zap_channel
you might want to add {ignore_early_media=true} to the dial string
and also make sure it was properly setup by testing for
sSession.ready() before executing intercept but I know that would work I
tested a similar situation in my box.
is the channel who is in the conference in there via a js also,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
I'm not familar with mod_openzap code nor with ss7boost stuff. I'm not
sure if the hack introduce some bad side effects but I know it seems to
solve this special problem. I can reproduce it any time.
Problem was, that when I make a call i
Hi Anthony,
Thanks again for your suggestion, unfortunately this doesn't do the
trick either.
The consumer (uuid) is indeed taking out of the conference call, but not
bridged to the customer_service_url.
I don't hear anything on the customer_service_url's phone. After a while
it gives me a r
wait a minute,
Can't you just do:
var cSession = new Session(customer_service_url);
cSession.execute("intercept", uuid);
intercept will steal the call from whatever it's doing and bridge it.
On Tue, Oct 28, 2008 at 8:50 AM, Birgit Arkesteijn <[EMAIL PROTECTED]>wrote:
> Hi Anthony,
>
> Sorry,
Are you sure that even does anything to help?
It says there not to change state to hangup when state is down or
terminating meaning it's already hanging up so how does removing the down do
any good, the state change from down to hangup would be vetoed anyway.
On Tue, Oct 28, 2008 at 8:21 AM, He
Hi Anthony,
Sorry, but it still doesn't work in
FreeSWITCH Version 1.0.trunk (597:10176)
It actually makes it worse sometimes.
Maybe it's a timing issue, but one of two things happen:
1.
The same problem happens as before:
The two calls are bridged, however the consumer remains in the
conference
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
File: mod_openzap.c
Function: channel_on_hangup
Line: 446
openzap was downloaded from "http://svn.openzap.org/svn/openzap/trunk/";
kind regards
helmut
Anthony Minessale schrieb:
> you forgot to say what line number and file that change was to.
Special thanks. Thatw as indeed the problem.
Klaus.
Original-Nachricht
> Datum: Mon, 27 Oct 2008 16:01:12 -0500
> Von: "Anthony Minessale" <[EMAIL PROTECTED]>
> An: freeswitch-users@lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] DTMF Star Event Inconsistent
> you clearl
you forgot to say what line number and file that change was to.
On Tue, Oct 28, 2008 at 4:31 AM, Helmut Kuper <[EMAIL PROTECTED]>wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> we use FS with mod_openzap to talk to a ss7boost server. we found that
> when we cancel a call
There are several settings in FS like max sessions, max sessions per second
(on the core) and max proceeding on sofia... The defaults are set fairly
reasonable but if you want to run very large volumes you¹ll need to crank
them up
From: jocke eriksson <[EMAIL PROTECTED]>
Reply-To:
Date: Tue, 2
Hello FS! I have a question about un earlier thread, the one where FS
returns a 503 response when ongoing calls reaches 240-250. I think Ron had
the same problem, so my question is, was the problem resolved? And if so,
what did he do?
___
Freeswitch-users
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
we use FS with mod_openzap to talk to a ss7boost server. we found that
when we cancel a call to PSTN befor media is established by FS the pstn
side won't be canceled as well. pstn side rings until timeout or voicebox.
I changed in mod_openzap.
Hello,
I still have problem after a fresh checkout.
I tried with MSVC++ 2008 express and I got the same errors too.
Tamas
Michael Jerris írta:
> This should work on a fresh checkout.
>
> Mike
>
> On Oct 27, 2008, at 10:23 AM, Tamas Cseke wrote:
>
>
>> Hello,
>>
>> I have a problem with window
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