Re: [Freeswitch-users] DTMF Event is not coming while using playback terminators.
Yes, I am using async mode only.. On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris 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 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 'COMMAND' > event. I have the following questions. > > > >Why the 'COMMAND' event came event filter is on? > >How to avoid this event in ESL? > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] DTMF Event is not coming while using playback terminators.
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 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? Thanks, Velusamy ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Flushing the Event buffer in Perl Event Socket
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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to stop the playback files
I am using async socket. Shall I set the regular expression for
palyback_terminators?
On Wed, Oct 28, 2009 at 2:26 PM, Brian West wrote:
> http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
>
> Are you using sync or async socket?
>
> /b
>
> On Oct 28, 2009, at 3:29 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-welcome_to_freeswitch.wav!ivr/
> > ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-
> > you_may_exit_by_hanging_up.wav");
> >
> > In a loop I am checking the DTMF event, if that event comes I
> > should stop the above palyback. How can I do it?
> >
> > Regards,
> > Velusamy.
> >
> > ___
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>
>
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[Freeswitch-users] How to stop the playback files
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-welcome_to_freeswitch.wav!ivr/ivr-this_ivr_will_let_you_test_features.wav!ivr/ivr-you_may_exit_by_hanging_up.wav");
In a loop I am checking the DTMF event, if that event comes I should stop
the above palyback. How can I do it?
Regards,
Velusamy.
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[Freeswitch-users] meaning of created_time channel variable.
Dear All, What is the value of created_time channel variable? Is this epoch seconds? Thanks & Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Perl event socket library problem
Dear All, I am implementing an IVR framework using Perl event socket libraries. At first I set socket mode async full. I have faced problem that the Perl statements executing before dailplan execution. So, I couldn't control my process. Next I have tried without async mode, it solved that race condition problem. I had another requirement that while executing external application I need to play some voice file some time limit. When I played that voice file in playback, the playback application blocked. So, I can't break that playback application after some time limit. I am hanging on this problem for past one week. Please any one help me to solve above problems... Thanks, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Asynchronous execution in ESL.pm
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. What is the problem? Please help me Thanks, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Play music on hold after parking the call
Any one please help me to solve the mentioned problem On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu wrote: > 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 > async mode in socket application? > > Thanks, > Velusamy. > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Play music on hold after parking the call
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 async mode in socket application? Thanks, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Difference between park and valet_park
Dear All, Could you please any one explain the difference between parking and valet parking? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.
Dear Sir, Please pardon me for this question. Now I corrected my mistake. It is working fine. Thank you vrey much for your valuable help.. On Fri, Sep 25, 2009 at 12:30 PM, Brian West wrote: > I beg to differ on that one.. what distro are you on? I use this all over > 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 is got. > My subroutine call is, > $conn->playAndGetDigits(1,1,1,8000,'#',"$play_list","ivr/ivr-please.wav","res"," > \\d+"); > > Is there any way to overcome this problem? > > Please help me... > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How do I get freeswitch variable which has DTMF values.
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 is got. My subroutine call is, $conn->playAndGetDigits(1,1,1,8000,'#',"$play_list","ivr/ivr-please.wav","res","\\d+"); Is there any way to overcome this problem? Please help me... On Fri, Sep 25, 2009 at 10:52 AM, Brian West 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 freeswitch variable I used getVar subroutine which is > > defined in ESL::IVR.pm file. When I print that digits, Perl program > > prints empty value while playing the menu itself. If I need to get > > the DTMF value I need to wait the perl program. > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How do I get freeswitch variable which has DTMF values.
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 defined in ESL::IVR.pm file. When I print that digits, Perl program prints empty value while playing the menu itself. If I need to get the DTMF value I need to wait the perl program. My question is why the Perl program prints the empty value before executing the getVar function? How can get the DTMF value by using getVar function without waiting in the Perl program? Please any one help me in this problem Thanks, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function
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 wrote:
> Use phrase macros as Brian said.
>
> On Thu, Sep 24, 2009 at 1:09 PM, Brian West wrote:
>
>> You can also use phrase macros. (and no its not just for TTS ;) )
>> /b
>>
>> On Sep 24, 2009, at 2:02 AM, Michael Collins wrote:
>>
>> Make sure that mod_file_string is built and loaded and then try the syntax
>> that is described here:
>> http://wiki.freeswitch.org/wiki/Mod_file_string#Examples
>>
>> Instead of a comma separated list you can use ! and be sure NOT to put a
>> space after the ! because the function delimits the arguments with spaces.
>> Try something like this:
>>
>> $conn->execute("play_and_get_digits", 1 1 1000 #
>> /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
>> /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
>>
>> Let us know how it goes.
>> -MC
>>
>>
>>
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>>
>
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[Freeswitch-users] How to play more than one voice file in play_and_get_digits function
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,
$conn->execute("play_and_get_digit", 1 1 1000 #
/usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav ,
/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
/usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
Is the above statement is right? Can I use more than one file in it?
OR
Can I use the play_get_digits as following?
$conn->execute("play_and_get_digit", 1 1 1000 #
/usr/local/freeswitch/en/us/callie/sounds/ivr/please.av
/usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
$conn->execute("play_and_get_digit", 1 1 1000 #
/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav
/usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+);
In the above statements how can I assure that the first application or
second application is executed?
When the digit is get while playing the first application the second
application should not be played. How Can I do that?
Is this my understanding wrong?
Correct me If I am wrong?
Please help me?
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[Freeswitch-users] Timeouts in Dial plan
Dear All, How to handle timeouts in Dialplan? Thanks, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Fwd: ALARM signal in esl libraries
Dear Greats, Could you please help me to solve this problem? -- Forwarded message -- From: velusamy velu 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 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 any idea to alive the connection after ALARM signal generation?? Please help me. Thanks, Velusamy ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ALARM signal in esl libraries
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 any idea to alive the connection after ALARM signal generation?? Please help me. Thanks, Velusamy ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] ESL for Perl
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': esl_wrap.cpp:(.text+0x1c10): undefined reference to `eslSetLogLevel' esl_wrap.o: In function `_wrap_ESLconnection_setEventLock': esl_wrap.cpp:(.text+0x3395): undefined reference to `ESLconnection::setEventLock(char const*)' esl_wrap.o: In function `_wrap_ESLconnection_setBlockingExecute': esl_wrap.cpp:(.text+0x3c65): undefined reference to `ESLconnection::setBlockingExecute(char const*)' esl_wrap.o: In function `_wrap_ESLconnection_execute': esl_wrap.cpp:(.text+0x41ab): undefined reference to `ESLconnection::execute(char const*, char const*, char const*)' I have understood the when creating soft link ESL.o that error has occurred. What is the problem? Is there any dependency to create that link? please help me... Thanks Regards, Velusamy ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Fwd: execute function in ESL.pm module is not working
L_EXECUTE] EVENT:[CHANNEL_EXECUTE_COMPLETE] EVENT:[DTMF] EVENT:[DTMF] EVENT:[DTMF] EVENT:[DTMF] IN Sigalarm--- In Play Digit Eneterd Digits=7485 Disconnected:1000 When alarm signal generated, it prints digits but it won't execute the "execute" function.. Please any one give suggestions where I made wrong... Thanks... Regards, Velusamy. On Thu, Aug 6, 2009 at 11:24 AM, Michael Collins wrote: > > > On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu > wrote: > >> 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 that we can take a look at them? > > Thanks, > MC > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Fwd: execute function in ESL.pm module is not working
Please any one help for this problem..
-- Forwarded message --
From: velusamy velu
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 signal in my Perl server program.
If ALARM signal occurred I execute the following statement in signal
handler.
"$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
The above statement didn't play that wave file. But before generating the
ALARM signal it worked.
What is the problem?
Please help me in this problem
Also Is there any idea to do timeout for DTMF digits?
Thanks...
Regards,
K.Velusamy
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[Freeswitch-users] execute function in ESL.pm module is not working
Dear All,
I registered alarm signal in my Perl server program.
If ALARM signal occurred I execute the following statement in signal
handler.
"$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
The above statement didn't play that wave file. But before generating the
ALARM signal it worked.
What is the problem?
Please help me in this problem
Also Is there any idea to do timeout for DTMF digits?
Thanks...
Regards,
K.Velusamy
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Re: [Freeswitch-users] Connecting FreeSWITCH with Asterisk
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... Thanks in Advance. Regards, Velusamy. On Tue, Jul 28, 2009 at 12:32 PM, Michael Collins wrote: > 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 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 > wrote: > >> 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 console >> >> "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to >> authenticate user "Velusamy" >> >> >;tag=69Q648H9NjrSK" >> >> I have read "Using Authentication" topic in the link, But I did understand >> that topic.. >> They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname >> I have to specify here? >> >> Please help me >> >> Regards, >> Velusamy.K >> >> ___ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Connecting FreeSWITCH with Asterisk
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 console "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" >;tag=69Q648H9NjrSK" I have read "Using Authentication" topic in the link, But I did understand that topic.. They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I have to specify here? Please help me Regards, Velusamy.K ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IAX configurations
I have loaded mod_iax now that error didn't come. But, When I call I have received following message in the console. "[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED" What configuration I missed? How to use sip to connect the Asterisk? Please give solutions above questions... __ Velusamy On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris 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. 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. > >field="destination_number" expression="^(222)$"> >data="iax/222:2...@192.168.6.94/$1"/> > > > * Next I have configured a 222 user in sip.conf > file at Asterisk machine. > * I wrote dialplan for that extension in > extension.conf file. > > When I tried to call 222 from FreeSWITCH, I have received following > error in Console. > "[ERR] switch_core_session.c:255 > switch_core_session_outgoing_channel() Could not locate channel type iax" > > What would be the problem? Is there any configuration I missed? > Please help me . > > Regards, > K.Velusamy. > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] IAX configurations
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. * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file. When I tried to call 222 from FreeSWITCH, I have received following error in Console. "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax" What would be the problem? Is there any configuration I missed? Please help me . Regards, K.Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] A stun server lookup
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() Failed to get external ip. I commented the stun configurations in vars.xml.conf file eventhough I am receiving the same error. Pleas any one give solution to solve this error Regards, Velusamy. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Creating a new User Agent
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, How to create new user agent ? How to relate the new user agent with sip internal profile ? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Problem in Adding another user in default directory
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Error in default Sofia profile checking
Dear Peter,
I have followed your steps, For me my FS and Twinkle running in
separate machine. But, I am still receiving the same error
"[ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO
CONTACT!"
Please give any suggestions to rectify this error..
Thanks in Advance,
Regards,
K.Velusamy.
>
>
> On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote:
>
> > I have several Twinkles running against freeswitch on a locally
> > installed machine (FS acts as a SIP/TLS proxy).
> > So in general Twinkle works (on various Ubuntu machines from 7 upto 9
> > with various Twinkle versions). It must be some kind of setting in
> > Twinkle. E.g.
> >
> >* set the local Twinkle SIP UDP port to 5062 in general settings
> >* Set the right network interface (e.g. eth0)
> >* In the profile do not set the realm
> >* Allow missing contact header on 200 OK
> >
> > Best regards
> > Peter
> >
> >
> >
> > 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 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 help me to solve this problem...
> >>>
> >>> Regards,
> >>> K.Velusamy.
> >>> ___
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users@lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> ___
> >> Freeswitch-users mailing list
> >> Freeswitch-users@lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > ___
> > Freeswitch-users mailing list
> > Freeswitch-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
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[Freeswitch-users] ERROR in Sofia internal profile
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 in Advance.. Regards, K.Velusamy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Error in default Sofia profile checking
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 help me to solve this problem...
Regards,
K.Velusamy.
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[Freeswitch-users] How to configure SIP(Sofia) profiles
Dear Friends, I am a newbie for FreeSWITCH. I was installed FreeSWITCH locally. I just wanted to test whether my FreeSWITCH is working fine. I need help from you that how to configure my Softphone(Twinkle) to use FreeSWITCH. I need steps to check my FreeSWITCH working with Twinkle. Please help me in this... Thanks in Advance. Regards, K.Velusamy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
