Hi,
I have tried google talk integration with FS and is working fine. Great Work!
Is it possible to have multiple concurrent incoming calls on the same
gmail account?
--
Regards,
Moiz Chinoy.
___
Freeswitch-users mailing list
*Hi Brian West,*
* I have installed the latest SVN Freeswitch trunk but still i get the same
error. How can i over come this problem.
2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java Framework
Loading...
2009-04-14 12:44:26 [ERR] modjava.c:133 load_config() Error loading
Baskar pisze:
*Hi Brian West,*
* I have installed the latest SVN Freeswitch trunk but still i get the
same error. How can i over come this problem.
2009-04-14 12:44:26 [NOTICE] modjava.c:244 mod_java_load() Java
Framework Loading...
2009-04-14 12:44:26 [ERR] modjava.c:133
*Hi,
I have installed latest java version jdk1.6.0 in this path
/usr/java/jdk1.6.0_04/bin
I have reconfigured FS ./configure --with-java=/usr/java/jdk1.6.0_04/bin,
make, make install
But when i run freeswitch in console i get this error.
2009-04-14 15:00:22 [ERR] modjava.c:133
Baskar pisze:
*Hi,
I have installed latest java version jdk1.6.0 in this path
/usr/java/jdk1.6.0_04/bin
I have reconfigured FS ./configure
--with-java=/usr/java/jdk1.6.0_04/bin, make, make install
But when i run freeswitch in console i get this error.
Exactly, configure java
Hi
even I have downloaded FreeSWITCH and using MS Visual Studio 2008. It is
building properly. I now want to configure it properly. Can you please give me
directions and also some links for good detailed comparisons between Asterisk
and FreeSWITCH and elaborative documentation for the same..
Hi,
My Java.conf.xml
configuration name=java.conf description=Java Plug-Ins
!-- Path to the Java 1.6 virtual machine to use --
javavm path=/usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so/
!-- Options to pass to Java --
options
!-- Your class path (make sure freeswitch.jar is on
Baskar pisze:
Hi,
My Java.conf.xml
configuration name=java.conf description=Java Plug-Ins
!-- Path to the Java 1.6 virtual machine to use --
javavm path=/usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so/
!-- Options to pass to Java --
options
!-- Your class path (make
*Hi,
I have not edited the java.conf.xml
*
*my libjvm.so file is loacted in this paths*
* [localhost ~]# locate libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so
/usr/lib/gcj-4.1.1/libjvm.so
Baskar pisze:
*Hi,
I have notice While reinstalling the Freeswitch i get this message While
both make and make install commands
making all mod_java
Note: src/org/freeswitch/Launcher.java uses unchecked or unsafe operations.
Note: Recompile with -Xlint:unchecked for details.
I am
http://wiki.freeswitch.org/wiki/Special:Search?search=asteriskgo=Go
On Apr 14, 2009, at 1:09 AM, kunal rao wrote:
Hi
even I have downloaded FreeSWITCH and using MS Visual Studio 2008.
It is building properly. I now want to configure it properly. Can
you please give me directions and also
Baskar pisze:
*Hi,
I have not edited the java.conf.xml
*
*my libjvm.so file is loacted in this paths*
* [localhost ~]# locate libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/client/libjvm.so
/usr/java/jdk1.6.0_04/jre/lib/i386/server/libjvm.so
/usr/lib/gcj-4.1.1/libjvm.so
yes, it should be.
On Tue, Apr 14, 2009 at 1:02 AM, Moiz Chinoy moizchi...@gmail.com wrote:
Hi,
I have tried google talk integration with FS and is working fine. Great
Work!
Is it possible to have multiple concurrent incoming calls on the same
gmail account?
--
Regards,
Moiz Chinoy.
try \d instead of \\d in your regex
On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola diego.vi...@gmail.com wrote:
Anthony,
I just tried to print the variable with the log app, with read it prints,
with play_and_get_digits doesn't.
I'm using latest SVN rev:
FreeSWITCH Version 1.0.trunk
Anthony,
Yes, it seems to work correct now. I did a couple of test calls, and tha audio
was good - thanks!
Another question about this scenario...
When doing a session.transfer(5000), this will transfer the call directly
into the dialplan without the use of loopback-channels. But that way
Hi Seven,
thanks a lot for the patch and all the Skypiax action.
I'm just back from Eastern vacations, let me clear the backlog and
I'll be back on this in a couple days.
Thanks again!
gm
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via
Hi UV,
seems a difficult one this one.
I have no much experience in RDP/terminal server.
If there is no way to have (or fake) audio driver on RDP/terminal
server apps, probably the Skype clients will not works (as you
experienced).
I'm sure, I've read it (:-) ), that Skype clients can be run
yes,
But if you plan is to bridge the call, the loopback channel is completely
unnecessary.
Be careful how much control you want =D getting a phone call up and running
is more work
than you think (see switch_ivr_originate.c)
On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson
Yes, I'm starting to realize that... :) but you to get everything right - if I
want to bridge a call, using the dialplan, then the only way is to use
loopback, right? If I don't want a loopback I'm able to bridge to the
destination directly?
//Peter
Från:
The bridge application will let you bridge right to a destination on
*another* box.
If you want to connect to a local extension like 5000 you can use the
transfer application or method.
session.transfer(5000);
exit();
or
session.execute(transfer, 5000);
exit();
On Tue, Apr 14, 2009 at 10:59
Hello everyone,
I'm trying to add Spanish support to say. I'm using something like:
include
language name=es sound-path=$${base_dir}/sounds/es/mx/asterisk
tts-engine=cepstral tts-voice=callie
X-PRE-PROCESS cmd=include data=demo/*.xml/ !-- Note: this
now grabs whole subdir, previously
Replying to myself... I forgot to indicate my version! I am running
trunk rev 12862 on CentOS 5 x86_64.
On Tue, Apr 14, 2009 at 12:51 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
I'm trying to add Spanish support to say. I'm using something like:
include
Nobody has written the es language files. Those would need to be
written.
/b
On Apr 14, 2009, at 11:54 AM, Kristian Kielhofner wrote:
Replying to myself... I forgot to indicate my version! I am running
trunk rev 12862 on CentOS 5 x86_64.
Brian West
br...@freeswitch.org
-- Meet us at
Allright - last question :) I'll try to be a little more specific. Lets say I
whant to do the following;
1. Dial into FreeSWITCH, to some kind of application (javascript or
whatever).
2. Answer that call, and let the user choose what to do; 1: record
message, 2: transfer to XXX
typically you would use transfer to the dest
then in the dialplan for you would
set hangup_after_bridge=true
try to call the phone
transfer back to your ivr
you can use channel variables to keep track of state.
On Tue, Apr 14, 2009 at 12:02 PM, Peter Olsson
The FreeSWITCH team is pleased to announce the immediate availability of
version 1.0.4pre4. Details are available here:
http://www.freeswitch.org/node/173
All are encouraged to upgrade as soon as possible.
Thanks to everyone for their feedback, ideas, and bug reports. Please keep
them coming.
KK,
Do you have someone who knows Spanish and who can translate? If not I will
whip up some volunteers from the FS community.
Thanks,
MC
On Tue, Apr 14, 2009 at 10:01 AM, Brian West br...@freeswitch.org wrote:
Nobody has written the es language files. Those would need to be written.
/b
On
I know spanish and I would translate it no problem. MC, get in touch
with me off-list so we can handle that.
I can also translate to portuguese-brazil.
jmesquita
On Apr 14, 2009, at 2:37 PM, Michael Collins wrote:
KK,
Do you have someone who knows Spanish and who can translate? If not
I
This also requires you to write all the phrase macros for voicemail,
ivr and other things in the demo in lang/en/
/b
On Apr 14, 2009, at 12:48 PM, João Mesquita wrote:
I know spanish and I would translate it no problem. MC, get in touch
with me off-list so we can handle that.
I can also
Brian,
For my application I just need to be able to say a string of numbers
- Caller ID, etc.
Other than the files used there is no syntax or grammar difference
(in Spanish) when compared to English. I should just be able to drop
the files in.
I'll have a problem when I need to handle
That works, thanks Anthm, you're the man.
Diego
On Tue, Apr 14, 2009 at 9:15 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try \d instead of \\d in your regex
On Tue, Apr 14, 2009 at 1:02 AM, Diego Viola diego.vi...@gmail.comwrote:
Anthony,
I just tried to print the variable
Hey guys,
If you need some Spanish help count with my help also.
Diego
On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins m...@freeswitch.org wrote:
Cool. We've had several volunteers start translating the phrase files into
Spanish and Brazilian Portugese. We'll keep you posted when we have
I'm a native spanish speaker, I can help too!
Nicolás Brenner
On Tue, Apr 14, 2009 at 2:56 PM, Diego Viola diego.vi...@gmail.com wrote:
Hey guys,
If you need some Spanish help count with my help also.
Diego
On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins m...@freeswitch.org wrote:
Cool. We've had several volunteers start translating the phrase files into
Spanish and Brazilian Portugese. We'll keep you posted when we have the
Spanish one ready. FYI, I committed a stub phrase_es.xml file but it
Hi all,
I'm looking for suggestions on which open source tools to use for creating
(or extending if there is already a project for this) a sip test suite.
I have already heard of sipp, but I want to know what others are using and
how they go about this before starting from scratch myself.
Some
BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px;
}Hello there,
In my previous encounter with FreeSwitch, I had found that Bret had
posted on the Mailing List somewhere about availability of the entire
FreeSwitch Wiki Documentation on a single PDF, this is useful coz at
the offset
It seems to me like the freeswitch platform itself would be a good place to
start. I haven't thoroughly thought this out, but maybe you could write a
test library using mod_language-of-your-choice designed to do human-like
things such as issuing dtmf tones, pausing, speaking, etc.
You could even
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