Yes, I am using async mode only..
On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote:
async?
On Nov 24, 2009, at 2:22 AM, velusamy velu wrote:
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set
Dear All,
I am using Perl ESL::IVR module to develop a simple IVR. I have
filtered DTMF events. I have also set playback_terminators to cut the
playback when giving the digits. I have faced problem that DTMF event has
not come if DTMF given while playing voice files. I have received
Dear All,
I receiving the events in while loop by using recvEventTimed method in
ESL.pm. I have to flush that Event buffer after some particular time. How
can I do it?
Thanks,
Velusamy
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Dear All,
I have played the list of voice files in playback like the following by
using ESL perl module,
$conn-execute(set,playback_delimiter=!);
$conn-execute(set,playback_sleep_val=100);
:29 AM, velusamy velu wrote:
Dear All,
I have played the list of voice files in playback like the
following by using ESL perl module,
$conn-execute(set,playback_delimiter=!);
$conn-execute(set,playback_sleep_val=100);
$conn-playback($sound_path.ivr/ivr
Dear All,
What is the value of created_time channel variable? Is this epoch
seconds?
Thanks Regards,
Velusamy.
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Dear All,
I have set my socket mode as full. I have used ESL.pm to develop an
IVR. I encounter the situation that I need play some music file while
executing some external application. So, I used executeAsync method to play
the music file. But the executeAsync application didn't work.
Any one please help me to solve the mentioned problem
On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu velu.techni...@gmail.comwrote:
Dear All,
I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I
Dear All,
Could you please any one explain the difference between parking and
valet parking?
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Dear All,
I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I executed
park after then I executed the playback the music didn't play.
Could any one please explain how can I solve this problem without using
+);
Is there any way to overcome this problem?
Please help me...
On Fri, Sep 25, 2009 at 10:52 AM, Brian West br...@freeswitch.org wrote:
Or use the socket without async so that it blocks till the action is
complete.
/b
On Sep 25, 2009, at 12:13 AM, velusamy velu wrote:
To get
the place and it works so i'm concerned about this.
/b
On Sep 25, 2009, at 1:55 AM, velusamy velu wrote:
Dear Sir,
If I disable the async mode in socket, the playAndGetDigits doesn't
exit after getting the DTMF value. It exit after time out seconds. But I
need to exit when DTMF digit
Dear All,
I am in the process of doing IVR development on FreeSWITCH. I am
having doubt in the play_and_get_digits application. I am using Perl
language for handling IVR.
How can I play more than one sound file in play_get_digits application?
For an example,
Dear All,
mod_file_string is working for me. Is there any problem by using
mod_file-string? If it so, how can use phrase macros in perl using ESL::IVR
module?
Please provide your valuable idea...
On Thu, Sep 24, 2009 at 10:25 PM, Diego Viola diego.vi...@gmail.com wrote:
Use phrase
Dear All,
I am doing IVR by using perl ESL libraries. I have used ESL::IVR
module. I get the DTMF by using playAndGetDigits subroutine which is defined
in ESL::IVR.pm. The DTMF digit stored in freeswitch digit variable.
To get the freeswitch variable I used getVar subroutine which is
Dear All,
I have registered ALARM signal in my perl program to handle the
DTMF digit timeout.
When ALARM signal generated the connection with ESL is automatically
closed.
I have checked the connection with connected: function, it returns 0.
Why the connection was closed?
Is there
Dear Greats,
Could you please help me to solve this problem?
-- Forwarded message --
From: velusamy velu velu.techni...@gmail.com
Date: Mon, Aug 10, 2009 at 12:08 PM
Subject: ALARM signal in esl libraries
To: freeswitch-users@lists.freeswitch.org
Dear All,
I have
Dear All,
How to handle timeouts in Dialplan?
Thanks,
Velusamy.
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Dear all,
When I do make for Perl ESL libraries in esl/perl directory I have got
the following error.
/usr/lib/gcc/i486-linux-gnu/4.1.2/../../../../lib/crt1.o: In function
`_start':
../sysdeps/i386/elf/start.S:115: undefined reference to `main'
esl_wrap.o: In function `_wrap_eslSetLogLevel':
at 11:38 PM, velusamy velu
velu.techni...@gmail.comwrote:
Please any one help for this problem..
Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
now so we'll ask for your patience... in the meantime could you pastebin
your script and your dialplan entry so
Dear All,
I registered alarm signal in my Perl server program.
If ALARM signal occurred I execute the following statement in signal
handler.
$conn-execute(playback,$sound_path.voicemail/vm-goodbye.wav)
The above statement didn't play that wave file. But before generating the
ALARM
Please any one help for this problem..
-- Forwarded message --
From: velusamy velu velu.techni...@gmail.com
Date: Wed, Aug 5, 2009 at 11:44 AM
Subject: execute function in ESL.pm module is not working
To: freeswitch-users@lists.freeswitch.org
Dear All,
I registered alarm
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
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 you make
a test call to the Asterisk box as that might give you some clue as to what
isn't working.
-MC
On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu
velu.techni
solutions above questions...
__
Velusamy
On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris m...@jerris.com wrote:
mod_iax isn't loaded. I suggest using sip anyways.
Mike
On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:
Dear All,
I have tried to call a Asterisk extension from FreeSWITCH
Dear All,
I have tried to call a Asterisk extension from FreeSWITCH. I have done
the following configurations,
* I have enabled mod_iax module in
modules.conf.xml file.
* Next I have configure following extension in
dialplan.
Dear All,
When I start the freeSWITCH, I am receiving the following errors,
2009-07-24 16:56:23 [ERR] sofia_glue.c:566
sofia_glue_ext_address_lookup() STUN Failed!
stun.freeswitch.org:3478[Remote Address Error!]
2009-07-24 16:56:23 [ERR] sofia.c:1972 config_sofia()
Dear All,
I want to create a new User Agent like sip configurations in
Asterisk. I checked default user agents 1000 to 1001. But I have bit
confused the relationship between default user agents and sip_profiles.
I need some help from you all for the following questions,
Dear All,
How to create another user agent like 1000 to 1919 in internal
profile.
Please provide some steps to do it..
Thanks in Advance,
Velusamy
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Mathieu Rene schrieb:
Chances are the registering UA didnt provide a Contact header
(required by rfc3261)
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 11-Jul-09, at 1:23 AM, velusamy velu wrote
Dear Friends,
When I reload the mod_sofia I have got the following error.
2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() Error
Creating SIP UA for profile: internal
Please any one explain about this error and please give any suggestions to
solve this problem..
Thanks
Dear Friends,
When I register my Softphone(Twinkle) with predefined sofia
registration(1000 with password 1234). I have got the following error
in FreeSWITCH console.
2009-07-11 09:37:16 [ERR] sofia_reg.c:1135
sofia_reg_handle_sip_i_register() NO CONTACT!
Please
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