From acrow at integrafin.co.uk  Sun Apr  1 12:40:23 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sun, 01 Apr 2012 09:40:23 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
Message-ID: <4F781477.2020602@integrafin.co.uk>

On 29/03/12 16:08, Peter Olsson wrote:
> ESL execute is async, so you will need to wait for the correct event to be returned, that tells you it's finished. EXECUTE_COMPLETE I think the event is called..
>
> /Peter
>
> 29 mar 2012 kl. 16:57 skrev "Alex Crow"<acrow at integrafin.co.uk>:
>

Peter,

Actually if execute is async the Wiki is wrong:

    "execute

    execute($app[, $arg][, $uuid])

    Execute a dialplan application, and wait for a response from the
    server. On socket connections not anchored to a channel (most of the
    time inbound), all three arguments are required -- $uuid specifies
    the channel to execute the application on.

    Returns an ESLevent object containing the response from the server.
    The getHeader("Reply-Text") method of this ESLevent object returns
    the server's response. The server's response will contain "+OK
    [Success Message]" on success or "-ERR [Error Message]" on failure.

    See the examples below for information on how to get a uuid to use
    when calling execute().
    executeAsync

    executeAsync($app[, $arg][, $uuid])

    Same as execute, but doesn't wait for a response from the server.

    This works by causing the underlying call to execute() to append
    "async: true" header in the message sent to the channel. "


As you can see there is a specific executeAsync call documented. Are 
they both actually async?

That aside, does this code look OK?

$esl = new ESLconnection('localhost','8021','ClueCon');

$event = $esl->events('plain', 'ALL');
$event = $esl->filter("Unique-ID",$uuid);

sleep (2);

$event = $esl->execute("play_and_get_digits","4 11 1 5000 # 
ivr/ivr-please_enter_the_phone_number.wav 
ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");

//$foo = $uuid . ' target_num';


while ($event->getType() != 'CHANNEL_EXECUTE_COMPLETE' ) {
                 $esl->execute("&sleep",100);
         }

$event=$esl->sendRecv("api uuid_getvar $uuid target_num");

Cheers

Alex

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From peter.olsson at visionutveckling.se  Sun Apr  1 13:33:27 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 1 Apr 2012 09:33:27 +0000
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F781477.2020602@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>,
	<4F781477.2020602@integrafin.co.uk>
Message-ID: <1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>

I believe that it's only sync in outbound mode. Maybe it would be easier to do what you want in outbound mode instead?

/Peter

________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Alex Crow [acrow at integrafin.co.uk]
Skickat: den 1 april 2012 10:40
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Dialplan/PHP esl variable problem.

On 29/03/12 16:08, Peter Olsson wrote:

ESL execute is async, so you will need to wait for the correct event to be returned, that tells you it's finished. EXECUTE_COMPLETE I think the event is called..

/Peter

29 mar 2012 kl. 16:57 skrev "Alex Crow" <acrow at integrafin.co.uk><mailto:acrow at integrafin.co.uk>:



Peter,

Actually if execute is async the Wiki is wrong:

"execute

execute($app[, $arg][, $uuid])

Execute a dialplan application, and wait for a response from the server. On socket connections not anchored to a channel (most of the time inbound), all three arguments are required -- $uuid specifies the channel to execute the application on.

Returns an ESLevent object containing the response from the server. The getHeader("Reply-Text") method of this ESLevent object returns the server's response. The server's response will contain "+OK [Success Message]" on success or "-ERR [Error Message]" on failure.

See the examples below for information on how to get a uuid to use when calling execute().
executeAsync

executeAsync($app[, $arg][, $uuid])

Same as execute, but doesn't wait for a response from the server.

This works by causing the underlying call to execute() to append "async: true" header in the message sent to the channel. "

As you can see there is a specific executeAsync call documented. Are they both actually async?

That aside, does this code look OK?

$esl = new ESLconnection('localhost','8021','ClueCon');

$event = $esl->events('plain', 'ALL');
$event = $esl->filter("Unique-ID",$uuid);

sleep (2);

$event = $esl->execute("play_and_get_digits","4 11 1 5000 # ivr/ivr-please_enter_the_phone_number.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");

//$foo = $uuid . ' target_num';


while ($event->getType() != 'CHANNEL_EXECUTE_COMPLETE' ) {
                $esl->execute("&sleep",100);
        }

$event=$esl->sendRecv("api uuid_getvar $uuid target_num");

Cheers

Alex

!DSPAM:4f7813f832766290219940!


From acrow at integrafin.co.uk  Sun Apr  1 13:50:37 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sun, 01 Apr 2012 10:50:37 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>,
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
Message-ID: <4F7824ED.7000807@integrafin.co.uk>



On 01/04/12 10:33, Peter Olsson wrote:
> I believe that it's only sync in outbound mode. Maybe it would be easier to do what you want in outbound mode instead?
>
> /Peter
>
> ________________________________
Thanks Peter,

I was thinking about that but since I'm not having much luck listening 
for events in inbound mode, I think I'd be running before I could walk 
(ie I'd have to listen for incoming calls rather than just using the 
dialplan to do that, and I don't want to risk breaking all the other 
calls on the system).

This isn't working for instance (after the execute):

while ($event->getType() != 'CHANNEL_EXECUTE_COMPLETE' ) {
                 $esl->execute("&sleep",1000);
         }

It actually seems to stop the call working completely, ie the caller 
doesn't hear the "enter the phone number". Not sure why, since that's 
executed before!

Cheers

Alex






From peter.olsson at visionutveckling.se  Sun Apr  1 14:26:18 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 1 Apr 2012 10:26:18 +0000
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F7824ED.7000807@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>,
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>,
	<4F7824ED.7000807@integrafin.co.uk>
Message-ID: <1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>

In your sample you're actually queuing "sleep(1000)" to the channel. So first it will queue play_and_get_digits, inside that loop it will find more commands in the queue, and exeute them, so in the end it will only execute a bunch of "sleep(1000)".

You're better of using a sleep in your module directly (not tell FS to sleep). Also you will need to poll the events in the loop somehow., checkout the perl sample here: http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Alex Crow [acrow at integrafin.co.uk]
Skickat: den 1 april 2012 11:50
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Dialplan/PHP esl variable problem.

On 01/04/12 10:33, Peter Olsson wrote:
> I believe that it's only sync in outbound mode. Maybe it would be easier to do what you want in outbound mode instead?
>
> /Peter
>
> ________________________________
Thanks Peter,

I was thinking about that but since I'm not having much luck listening
for events in inbound mode, I think I'd be running before I could walk
(ie I'd have to listen for incoming calls rather than just using the
dialplan to do that, and I don't want to risk breaking all the other
calls on the system).

This isn't working for instance (after the execute):

while ($event->getType() != 'CHANNEL_EXECUTE_COMPLETE' ) {
                 $esl->execute("&sleep",1000);
         }

It actually seems to stop the call working completely, ie the caller
doesn't hear the "enter the phone number". Not sure why, since that's
executed before!

Cheers

Alex





_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4f7824ba32766551348335!



From mytemike72 at gmail.com  Sun Apr  1 16:05:30 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Sun, 1 Apr 2012 14:05:30 +0200
Subject: [Freeswitch-users] Api function to check if media is currently
	being played
Message-ID: <001a01cd0fff$bcb75120$3625f360$@com>

Hi All,

 

I am using an remote host to control my parked call using an inbound socket
connection.

I am sending  "uuid_broadcast" to play an audio file to the call. Before it
is being bridged. Everything works ok.

 

The problem is I need to continue my code (ESL) to instruct the bridge etc,
but I need to do that when the audio file being played is finished. 

 

Is there an api function for this? 

 

I tried "uuid_buglist" to check if that would return me some useful  data,
but that doesn't seem to be working (at least for what I need ..)

Is the something like "uuid_mediaReady" ?  show api didn't show me something
similar..

 

 

Thanks!,

Mike.

 

Ps. I believe tricking with checking file length of al wavs and calculating
the duration upfront would not be the way doing this. ;-)

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From peter.olsson at visionutveckling.se  Sun Apr  1 17:00:39 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 1 Apr 2012 13:00:39 +0000
Subject: [Freeswitch-users] Api function to check if media is
	currently	being played
In-Reply-To: <001a01cd0fff$bcb75120$3625f360$@com>
References: <001a01cd0fff$bcb75120$3625f360$@com>
Message-ID: <1FFF97C269757C458224B7C895F35F15084E31@cantor.std.visionutv.se>

You will need to poll the events coming in. When the file playback is finished it will return an event.

To make it easy, just log all events coming in to start with, and after that try to handle it accordingly.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michael Lutz [mytemike72 at gmail.com]
Skickat: den 1 april 2012 14:05
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] Api function to check if media is currently being played

Hi All,

I am using an remote host to control my parked call using an inbound socket connection.
I am sending  ?uuid_broadcast? to play an audio file to the call. Before it is being bridged. Everything works ok.

The problem is I need to continue my code (ESL) to instruct the bridge etc, but I need to do that when the audio file being played is finished.

Is there an api function for this?

I tried ?uuid_buglist? to check if that would return me some useful  data, but that doesn?t seem to be working (at least for what I need ..)
Is the something like ?uuid_mediaReady? ?  show api didn?t show me something similar..


Thanks!,
Mike.

Ps. I believe tricking with checking file length of al wavs and calculating the duration upfront would not be the way doing this? ;-)
!DSPAM:4f78440532761728310852!


From mytemike72 at gmail.com  Sun Apr  1 17:16:53 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Sun, 1 Apr 2012 15:16:53 +0200
Subject: [Freeswitch-users] Api function to check if media
	is	currently	being played
In-Reply-To: <1FFF97C269757C458224B7C895F35F15084E31@cantor.std.visionutv.se>
References: <001a01cd0fff$bcb75120$3625f360$@com>
	<1FFF97C269757C458224B7C895F35F15084E31@cantor.std.visionutv.se>
Message-ID: <001f01cd1009$b5c466f0$214d34d0$@com>

Peter,

Thanks, yes I know how to handle the events, I was just wondering as I would
expect such function to exist. Because I can imagine situations where the
media which is being played is initiated by an (other) external connection
(or the session) and does not always knows what or how many PLAYBACK_STOPS
it should wait for... 

But for now, for me this will do!,

Thanks for your response.

Regards,
Mike.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: zondag 1 april 2012 15:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Api function to check if media is currently
being played

You will need to poll the events coming in. When the file playback is
finished it will return an event.

To make it easy, just log all events coming in to start with, and after that
try to handle it accordingly.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f?r Michael Lutz
[mytemike72 at gmail.com]
Skickat: den 1 april 2012 14:05
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] Api function to check if media is currently being
played

Hi All,

I am using an remote host to control my parked call using an inbound socket
connection.
I am sending  ?uuid_broadcast? to play an audio file to the call. Before it
is being bridged. Everything works ok.

The problem is I need to continue my code (ESL) to instruct the bridge etc,
but I need to do that when the audio file being played is finished.

Is there an api function for this?

I tried ?uuid_buglist? to check if that would return me some useful  data,
but that doesn?t seem to be working (at least for what I need ..)
Is the something like ?uuid_mediaReady? ?  show api didn?t show me something
similar..


Thanks!,
Mike.

Ps. I believe tricking with checking file length of al wavs and calculating
the duration upfront would not be the way doing this
 ;-)
!DSPAM:4f78440532761728310852!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From peter.olsson at visionutveckling.se  Sun Apr  1 18:41:09 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 1 Apr 2012 14:41:09 +0000
Subject: [Freeswitch-users] Api function to check if
	media	is	currently	being played
In-Reply-To: <001f01cd1009$b5c466f0$214d34d0$@com>
References: <001a01cd0fff$bcb75120$3625f360$@com>
	<1FFF97C269757C458224B7C895F35F15084E31@cantor.std.visionutv.se>,
	<001f01cd1009$b5c466f0$214d34d0$@com>
Message-ID: <1FFF97C269757C458224B7C895F35F1508509D@cantor.std.visionutv.se>

For playback it's also possible to set your own local variables, which will be available on the PLAYBACK_STOP event. It's also possible to use the scoped variables, they are visible in CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE - for that specific execution only.

See http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback#Example_for_specific_playback_variables and http://wiki.freeswitch.org/wiki/Channel_Variables#Scoped_Variables for more information.

With the help of this you can make sure you only handle the right events, and not events created from other applications.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Lutz [mytemike72 at gmail.com]
Skickat: den 1 april 2012 15:16
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] Api function to check if media     is      currently       being played

Peter,

Thanks, yes I know how to handle the events, I was just wondering as I would
expect such function to exist. Because I can imagine situations where the
media which is being played is initiated by an (other) external connection
(or the session) and does not always knows what or how many PLAYBACK_STOPS
it should wait for...

But for now, for me this will do!,

Thanks for your response.

Regards,
Mike.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: zondag 1 april 2012 15:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Api function to check if media is currently
being played

You will need to poll the events coming in. When the file playback is
finished it will return an event.

To make it easy, just log all events coming in to start with, and after that
try to handle it accordingly.

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f?r Michael Lutz
[mytemike72 at gmail.com]
Skickat: den 1 april 2012 14:05
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] Api function to check if media is currently being
played

Hi All,

I am using an remote host to control my parked call using an inbound socket
connection.
I am sending  ?uuid_broadcast? to play an audio file to the call. Before it
is being bridged. Everything works ok.

The problem is I need to continue my code (ESL) to instruct the bridge etc,
but I need to do that when the audio file being played is finished.

Is there an api function for this?

I tried ?uuid_buglist? to check if that would return me some useful  data,
but that doesn?t seem to be working (at least for what I need ..)
Is the something like ?uuid_mediaReady? ?  show api didn?t show me something
similar..


Thanks!,
Mike.

Ps. I believe tricking with checking file length of al wavs and calculating
the duration upfront would not be the way doing this? ;-)


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f7854a532765776788417!



From acrow at integrafin.co.uk  Sun Apr  1 20:29:51 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sun, 01 Apr 2012 17:29:51 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>,
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>,
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
Message-ID: <4F78827F.7020401@integrafin.co.uk>

On 01/04/12 11:26, Peter Olsson wrote:
> In your sample you're actually queuing "sleep(1000)" to the channel. So first it will queue play_and_get_digits, inside that loop it will find more commands in the queue, and exeute them, so in the end it will only execute a bunch of "sleep(1000)".
>
> You're better of using a sleep in your module directly (not tell FS to sleep). Also you will need to poll the events in the loop somehow., checkout the perl sample here: http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example

Peter,

I've changed to the below (adapted that Perl example) and I only hear 
the prompt for digits when the webserver times out after 60 seconds. Any 
time I make any call to ESL after the execute, I don't hear the prompt 
in a timely fashion. This is so complicated as I'm not really that good 
at software development, I'm more a systems/networks guy. I feel like a 
total thickie here, but the docs don't seem to explain in which order I 
should call things and expect to get replies, etc.

$esl = new ESLconnection('localhost','8021','ClueCon');

$event = $esl->events('plain', 'ALL');
//$event = $esl->filter("Unique-ID",$uuid);

sleep (1);

$res = $esl->execute("play_and_get_digits","4 11 1 5000 # 
ivr/ivr-please_enter_the_phone_number.wav 
ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");

$job_uuid = $res->getHeader("Job-UUID");
$status = $res->getHeader("Reply-Text");



//$foo = $uuid . ' target_num';

echo $status;

$stay_connected = 1;
while ( $stay_connected ) {
     $e = $esl->recvEventTimed(0);
     if ( $e ) {
         $ev_name = $e->getHeader("Event-Name");
         # Should we check for the $job_uuid to match the background job ?
         //if ( $ev_name == 'BACKGROUND_JOB' ) {
         //    if ( $e->getHeader("Job-UUID") == $uuid ) {
         //        $call_result = $e->getBody();
         //        echo "Result of call to $target was $call_result\n\n";
         //    }
         //} else
         echo $ev_name;
         if ( $ev_name == 'EXECUTE_COMPLETE' ) {
             //my $digit = $e->getHeader("DTMF-Digit");
            // print "Received DTMF digit: $digit\n";
             //if ( $digit =~ m/\D/ ) {a
              //   print "Exiting...\n";
                 //if ( $e->getHeader("Job-UUID") == $job_uuid ) {
                 $stay_connected = 0;
                 //}
             //}
         } else {
       # Some other event
         }
     } else {
     # do other things while waiting for events
         //sleep (1);
     }
}

$event=$esl->sendRecv("api uuid_getvar $uuid target_num");


$ret = $event->getBody();

$esl->disconnect();

Apologies for my newbishness and many thanks for your help so far.

Hoping for an epiphany when I sudddenly understand what I am doing!

Alex


From lazyvirus at gmx.com  Sun Apr  1 20:45:18 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sun, 1 Apr 2012 18:45:18 +0200
Subject: [Freeswitch-users] VM sound very choppy
In-Reply-To: <1FFF97C269757C458224B7C895F35F1507CB41@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1507CB41@cantor.std.visionutv.se>
Message-ID: <20120401184518.4bbdd1da@anubis.defcon1>

On Fri, 23 Mar 2012 07:24:30 +0000
Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Another case of virtual machine setup maybe,
> or is this real hardware?

Thanks for having pointed the hardware, Peter.

In the past I had some small troubles (and high frequency noise)
coming from the motherboard Ethernet adapter (a via_rhine chipset)
but after a kernel change, everything seemed fine, except the noise.

Today I was listening to a web radio and lost it, it also happened I
lost all connectivity!  Only an ifdown eth0; ifup eth0 recovered.

So, I'm now pretty sure this problem of choppy sound is coming from
the Ethernet adapter (may be it could also be the drv, but I didn't
find anything about that on the web) - it took me a while to
realize that because this in not a very common failure.

So, I'm gonna deactivate this MB card and insert a good one.

JY
-- 


From lazyvirus at gmx.com  Sun Apr  1 22:57:14 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sun, 1 Apr 2012 20:57:14 +0200
Subject: [Freeswitch-users] moh is not transcoded for some CODECs
Message-ID: <20120401205714.1bfb2111@anubis.defcon1>

Hi list,

I'm playing with the FS original diaplan and noticed something
strange, when twinkle has G711(a|u) or GSM into its list the moh's
played, but when I only leave iLBC & SPEEX it fails.

I modified vars.xml and added iLCB at 30i on both global_codec_prefs &
outbound_codec_prefs, so AFAI understand how FS works it should be Ok.
But the log shows comparisons only against G7221s, PCMU, PCMA & GSM:

2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G7221:115:32000:20:48000]
2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G7221:107:16000:20:32000]
2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[G722:9:8000:20:64000]
2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[PCMU:0:8000:20:64000]
2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/[PCMA:8:8000:20:64000]
2012-04-01 20:52:21.230467 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [iLBC:96:8000:30:13330]/
[GSM:3:8000:20:13200]

does MOH only work with these codecs, or did I miss another place
than vars.xml to set the MOH available CODECs?

JY
-- 
"Surely you can't be serious."
"I am serious, and stop calling me Shirley." -- "Airplane"


From lazyvirus at gmx.com  Sun Apr  1 23:14:09 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sun, 1 Apr 2012 21:14:09 +0200
Subject: [Freeswitch-users] moh is not transcoded for some CODECs
 [SOLVED]
In-Reply-To: <20120401205714.1bfb2111@anubis.defcon1>
References: <20120401205714.1bfb2111@anubis.defcon1>
Message-ID: <20120401211409.54b8b93b@anubis.defcon1>

On Sun, 1 Apr 2012 20:57:14 +0200
Bzzz <lazyvirus at gmx.com> wrote:

> 
> I'm playing with the FS original diaplan and noticed something
> strange, when twinkle has G711(a|u) or GSM into its list the moh's
> played, but when I only leave iLBC & SPEEX it fails.

Yeah well, FS prefers a lot more iLBC at 30i rather than iBLC at 30i :(

Sornoise for the ry

JY
-- 



From xieweikai at gmail.com  Sun Apr  1 09:43:58 2012
From: xieweikai at gmail.com (Weikai Xie)
Date: Sun, 1 Apr 2012 13:43:58 +0800
Subject: [Freeswitch-users] Commands to show RTP statistics
Message-ID: <CAHk1ir+ToEj81RKKQrrdReAnzKrrKCk4Bsz68Ug_Tku_jgqs8A@mail.gmail.com>

Hi,
    Is there any commands or means in Freeswitch which can be used to show
the recent send/recv RTP statistics, such as packets loss rate, bandwidth,
etc, on a channel?
    I've tried look for related commands in the wiki but can't get a clue.

Regards,

Weikai Xie
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From standa.kutil at gmail.com  Sun Apr  1 23:25:03 2012
From: standa.kutil at gmail.com (Stanislav Kutil)
Date: Sun, 1 Apr 2012 21:25:03 +0200
Subject: [Freeswitch-users] no-way audio when accepting a call with iLBC
Message-ID: <B2E5927C-996C-4F92-B66D-1D88A7039FD8@gmail.com>

I'm experiencing no-way audio when using iLBC codec with FS.

interesting snippet from the dump:

we receive this INVITE, note iLBC is included both as payload 98 and payload 102.

INVITE sip:1203 at 192.168.1.245:4956;rinstance=018EE99B SIP/2.0
Via: SIP/2.0/UDP 93.185.48.4;rport;branch=z9hG4bKSvyXBN2p9cZNp
Route: <sip:1203 at 78.80.18.10:10142>;rinstance=018EE99B
Max-Forwards: 69
From: "Extension 1205" <sip:1205 at pbx.acrobits.cz>;tag=43Bm32867KUBS
To: <sip:1203 at 192.168.1.245:4956;rinstance=018EE99B>
Call-ID: 2ec155bb-f612-122f-3292-001a929b68ca
CSeq: 26276632 INVITE
Contact: <sip:mod_sofia at 93.185.48.4:5060>
User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-a239914 2012-03-26 10-53-39 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 486
X-FS-Support: update_display,send_info
Remote-Party-ID: "Extension 1205" <sip:1205 at pbx.acrobits.cz>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1333204589 1333204590 IN IP4 93.185.48.4
s=FreeSWITCH
c=IN IP4 93.185.48.4
t=0 0
m=audio 20804 RTP/AVP 0 8 9 98 3 99 100 102 101 13
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:99 G7221/32000
a=fmtp:99 bitrate=48000
a=rtpmap:100 G7221/16000
a=fmtp:100 bitrate=32000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 22668 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 H264/90000


we pick 102 version:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.185.48.4;rport=5060;branch=z9hG4bKSvyXBN2p9cZNp;received=93.185.48.4
Contact: <sip:1203 at 192.168.1.245:4956>
From: "Extension 1205" <sip:1205 at pbx.acrobits.cz>;tag=43Bm32867KUBS
Call-ID: 2ec155bb-f612-122f-3292-001a929b68ca
CSeq: 26276632 INVITE
To: <sip:1203 at 192.168.1.245:4956;rinstance=018EE99B>;tag=8466C9086B8AF5AE5586CEC0F3CFB45C
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Content-Type: application/sdp
Content-Length: 307

v=0
o=- 03870 51862 IN IP4 10.39.33.21
s=tinvgxd
c=IN IP4 10.39.33.21
t=0 0
m=audio 51132 RTP/AVP 102 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=rtpmap:102 ILBC/8000
a=fmtp:101 0-15
a=fmtp:102 mode=20
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34 103
a=rtpmap:103 H264/90000
a=rtpmap:34 H263/90000


the NAT traversal feature of FS to send audio back to where it is send does not kick in in this case so the iLBC stream from us is ignored by FS and iLBC stream from FS goes to some private IP (we are not on the same private network as FS)

1) as soon as we send a re-INVITE it fixes itself (now only 102 is included)
2) If we declare the correct public IP in the SDP it also works.
3) other codecs work fine


So to me the problem can either be the dual payload, or iLBC vs ILBC. I think the latter is correct btw.

Thanks,

Stan
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From bdfoster at endigotech.com  Mon Apr  2 00:20:29 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 1 Apr 2012 16:20:29 -0400
Subject: [Freeswitch-users] Drive by DTMF
Message-ID: <CAKV-jFy0RpVVhP2__OiMVhJQbvaG1GjKarFwngELVtZkwgSG6A@mail.gmail.com>

INDIANAPOLIS, INDIANA - EndigoVoIP, a division of Endigo Communications,
has released a module for the open source telephony toolkit FreeSWITCH that
gives a user the ability to control a full sized automobile using the
buttons found on any telephone and a live video feed.

Mod_drivebydtmf, currently in beta status, will give the user to control
almost every function of a car. Keys one and three are used for left and
right turn signals, keys 2 and 8 will be used for acceleration and braking,
and keys 4 and 6 will be used to turn the vehicle left and right. The
module will also use the star key for an IVR that will be used to control
various other functions of the car, including gear selection, locking and
unlocking the doors on the car, and rolling down and up the power windows.
The pound (hash) key will be used as a panic button, which will stop the
car, place the car in park, shut off the engine, lock the doors, and call
emergency services.

Although the company says the module is in beta, it encourages
telecommunications administrators to deploy the module on a production
system. "There shouldn't be many bugs with the module, although we have
witnessed a few accidents due to the car not applying the brakes," says CEO
Brian Foster, "but we believe the error was due to a misconfiguration of
the module or driver error."

The company expects to see this the module used in a wide variety of
applications, from controlling UPS trucks on their delivery route to
driving the biggest military tanks on earth.

According to sources, the company expects to come out of beta by the end of
the year. And of course, EndigoVoIP will release the module as open source
under a Mozilla license.

-BDF
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From asannucci at gmail.com  Mon Apr  2 00:27:36 2012
From: asannucci at gmail.com (Bakko)
Date: Sun, 01 Apr 2012 22:27:36 +0200
Subject: [Freeswitch-users] Drive by DTMF
In-Reply-To: <CAKV-jFy0RpVVhP2__OiMVhJQbvaG1GjKarFwngELVtZkwgSG6A@mail.gmail.com>
References: <CAKV-jFy0RpVVhP2__OiMVhJQbvaG1GjKarFwngELVtZkwgSG6A@mail.gmail.com>
Message-ID: <4F78BA38.2060108@gmail.com>

I need this module :)

El 01/04/2012 22:20, Brian Foster escribi?:
>
> INDIANAPOLIS, INDIANA - EndigoVoIP, a division of Endigo 
> Communications, has released a module for the open source telephony 
> toolkit FreeSWITCH that gives a user the ability to control a full 
> sized automobile using the buttons found on any telephone and a live 
> video feed.
>
> Mod_drivebydtmf, currently in beta status, will give the user to 
> control almost every function of a car. Keys one and three are used 
> for left and right turn signals, keys 2 and 8 will be used for 
> acceleration and braking, and keys 4 and 6 will be used to turn the 
> vehicle left and right. The module will also use the star key for an 
> IVR that will be used to control various other functions of the car, 
> including gear selection, locking and unlocking the doors on the car, 
> and rolling down and up the power windows. The pound (hash) key will 
> be used as a panic button, which will stop the car, place the car in 
> park, shut off the engine, lock the doors, and call emergency services.
>
> Although the company says the module is in beta, it encourages 
> telecommunications administrators to deploy the module on a production 
> system. "There shouldn't be many bugs with the module, although we 
> have witnessed a few accidents due to the car not applying the 
> brakes," says CEO Brian Foster, "but we believe the error was due to a 
> misconfiguration of the module or driver error."
>
> The company expects to see this the module used in a wide variety of 
> applications, from controlling UPS trucks on their delivery route to 
> driving the biggest military tanks on earth.
>
> According to sources, the company expects to come out of beta by the 
> end of the year. And of course, EndigoVoIP will release the module as 
> open source under a Mozilla license.
>
> -BDF
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Mon Apr  2 00:29:08 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 1 Apr 2012 23:29:08 +0300
Subject: [Freeswitch-users] Commands to show RTP statistics
In-Reply-To: <CAHk1ir+ToEj81RKKQrrdReAnzKrrKCk4Bsz68Ug_Tku_jgqs8A@mail.gmail.com>
References: <CAHk1ir+ToEj81RKKQrrdReAnzKrrKCk4Bsz68Ug_Tku_jgqs8A@mail.gmail.com>
Message-ID: <CALNZuwmNRae+fHR=G1w4raOzm8BkEp4BL5JYr3=5Day90ngt6g@mail.gmail.com>

Check: http://wiki.freeswitch.org/wiki/RTCP
And/or: http://www.voipmonitor.org/

(Best I got for now.)
-Avi


On Sun, Apr 1, 2012 at 8:43 AM, Weikai Xie <xieweikai at gmail.com> wrote:

> Hi,
>     Is there any commands or means in Freeswitch which can be used to show
> the recent send/recv RTP statistics, such as packets loss rate, bandwidth,
> etc, on a channel?
>     I've tried look for related commands in the wiki but can't get a clue.
>
> Regards,
>
> Weikai Xie
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From daggelinckxmichel at gmail.com  Mon Apr  2 03:17:35 2012
From: daggelinckxmichel at gmail.com (Michel Daggelinckx)
Date: Mon, 02 Apr 2012 01:17:35 +0200
Subject: [Freeswitch-users] Drive by DTMF
In-Reply-To: <4F78BA38.2060108@gmail.com>
References: <CAKV-jFy0RpVVhP2__OiMVhJQbvaG1GjKarFwngELVtZkwgSG6A@mail.gmail.com>
	<4F78BA38.2060108@gmail.com>
Message-ID: <4F78E20F.3000303@gmail.com>

Or a calendar


On 04/01/2012 10:27 PM, Bakko wrote:
> I need this module :)
>
> El 01/04/2012 22:20, Brian Foster escribi?:
>>
>> INDIANAPOLIS, INDIANA - EndigoVoIP, a division of Endigo 
>> Communications, has released a module for the open source telephony 
>> toolkit FreeSWITCH that gives a user the ability to control a full 
>> sized automobile using the buttons found on any telephone and a live 
>> video feed.
>>
>> Mod_drivebydtmf, currently in beta status, will give the user to 
>> control almost every function of a car. Keys one and three are used 
>> for left and right turn signals, keys 2 and 8 will be used for 
>> acceleration and braking, and keys 4 and 6 will be used to turn the 
>> vehicle left and right. The module will also use the star key for an 
>> IVR that will be used to control various other functions of the car, 
>> including gear selection, locking and unlocking the doors on the car, 
>> and rolling down and up the power windows. The pound (hash) key will 
>> be used as a panic button, which will stop the car, place the car in 
>> park, shut off the engine, lock the doors, and call emergency services.
>>
>> Although the company says the module is in beta, it encourages 
>> telecommunications administrators to deploy the module on a 
>> production system. "There shouldn't be many bugs with the module, 
>> although we have witnessed a few accidents due to the car not 
>> applying the brakes," says CEO Brian Foster, "but we believe the 
>> error was due to a misconfiguration of the module or driver error."
>>
>> The company expects to see this the module used in a wide variety of 
>> applications, from controlling UPS trucks on their delivery route to 
>> driving the biggest military tanks on earth.
>>
>> According to sources, the company expects to come out of beta by the 
>> end of the year. And of course, EndigoVoIP will release the module as 
>> open source under a Mozilla license.
>>
>> -BDF
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Mon Apr  2 03:53:55 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 1 Apr 2012 19:53:55 -0400
Subject: [Freeswitch-users] Drive by DTMF
In-Reply-To: <4F78E20F.3000303@gmail.com>
References: <CAKV-jFy0RpVVhP2__OiMVhJQbvaG1GjKarFwngELVtZkwgSG6A@mail.gmail.com>
	<4F78BA38.2060108@gmail.com> <4F78E20F.3000303@gmail.com>
Message-ID: <CAKV-jFzjMSXgJEeqS4So+mbM3a_taHdwe6Nk-peWhtOBU-B-wg@mail.gmail.com>

Oh crap, guess I should have released it tomorrow...

On Sun, Apr 1, 2012 at 7:17 PM, Michel Daggelinckx <
daggelinckxmichel at gmail.com> wrote:

>  Or a calendar
>
>
>
> On 04/01/2012 10:27 PM, Bakko wrote:
>
> I need this module :)
>
> El 01/04/2012 22:20, Brian Foster escribi?:
>
> INDIANAPOLIS, INDIANA - EndigoVoIP, a division of Endigo Communications,
> has released a module for the open source telephony toolkit FreeSWITCH that
> gives a user the ability to control a full sized automobile using the
> buttons found on any telephone and a live video feed.
>
> Mod_drivebydtmf, currently in beta status, will give the user to control
> almost every function of a car. Keys one and three are used for left and
> right turn signals, keys 2 and 8 will be used for acceleration and braking,
> and keys 4 and 6 will be used to turn the vehicle left and right. The
> module will also use the star key for an IVR that will be used to control
> various other functions of the car, including gear selection, locking and
> unlocking the doors on the car, and rolling down and up the power windows.
> The pound (hash) key will be used as a panic button, which will stop the
> car, place the car in park, shut off the engine, lock the doors, and call
> emergency services.
>
> Although the company says the module is in beta, it encourages
> telecommunications administrators to deploy the module on a production
> system. "There shouldn't be many bugs with the module, although we have
> witnessed a few accidents due to the car not applying the brakes," says CEO
> Brian Foster, "but we believe the error was due to a misconfiguration of
> the module or driver error."
>
> The company expects to see this the module used in a wide variety of
> applications, from controlling UPS trucks on their delivery route to
> driving the biggest military tanks on earth.
>
> According to sources, the company expects to come out of beta by the end
> of the year. And of course, EndigoVoIP will release the module as open
> source under a Mozilla license.
>
> -BDF
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
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From wesleyakio at tuntscorp.com  Mon Apr  2 04:41:08 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Sun, 1 Apr 2012 21:41:08 -0300
Subject: [Freeswitch-users] Cisco UCM 7.1 3pcc hold problem
Message-ID: <CAGTsDSs0xCGYtCkFsGZmaiFT4HQOf9txz7NA5THDeG11Ubr=Ng@mail.gmail.com>

Hi all,

I'm trying SIP Trunk between CUCM 7.1 and FS. Unfortunately MTP is not an
option, what leaves me with a whole lot of problems.

The one I'm currently banging my head with is hold.

I do not have the traces here, I'll have my hands on them tomorrow...
Meanwhile what happens is the following:

CISCO -> FS

->INVITE - no SDP
<-Trying
<-Ringing
<-200 - SDP
->ACK - SDP

At this point the call is setup and things work as they should.

Cisco tries hold with freeswitch

->INVITE - SDP with IP 0.0.0.0
<-200 - SPD as usual
-> ACK - no SDP

the call is On hold, freeswitch is playing MOH

microseconds later, cisco tries to recover the call and connect to its MOH
server:

-> INVITE - no SDP
<- 200 - SDP as usual
-> ACK - Valid SDP pointing to a moh server

At this point freeswitch continues peacefully to play it's own MOH
Cisco is sending RTP but FS is ignoring it
>From this point on the call is lost and will remain on MOH until someone
hangs up... any atempt to recover the call fails.

I'm sorry for the lack of traces as pcaps, i'll add the tomorrow...

Have someone gone through this problem or have any idea on what can i try?

Best,

Wesley Akio
TuntsCorp.com
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From philq at qsystemsengineering.com  Mon Apr  2 08:37:19 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Mon, 02 Apr 2012 00:37:19 -0400
Subject: [Freeswitch-users] Possible registration/presence bug - Sometimes
 get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
Message-ID: <000301cd108a$4d3f55a0$e7be00e0$@com>

Hi,

I've noticed what appears to be a bug which I wanted to bring to your
attention.  Sometimes when FS is started, extensions can't be called even
after they've registered.  This doesn't happen all the time. and restarting
FS always solves the problem.  Just to be clear - this problem only
sometimes occurs when FS has been started/restarted, like after an update.
Once FS has "started successfully" and sees presence for registered
extensions, it continues to work fine.

It's all or nothing - no extensions can be dialed when this issue is
present, whether they're local or NATted.  If one extension can be reached,
you can reach them all (if they're registered, of course).

See debug info below on an attempt to dial a local (no NAT) extension - this
was one of the recent beta git pulls and the problem has been present
through several updates.  Let me know if you need any additional info.


freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostn
ame
102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;tra
nsport=udp,1333338853,192.168.1.4,5060,udp,server.home
257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.182:5060;t
ransport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;t
ransport=udp,1333338743,192.168.1.4,5060,udp,server.home
227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.222.182:1066;t
ransport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.222.182:1068;t
ransport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.222.182:1067;t
ransport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.100.94:5462
9;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.100.94:6480
;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.22
2.182:1062,1333338779,74.93.xx.xx,1062,udp,server.home

9 total.

2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
acl "domains". Falling back to Digest auth.
2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
(INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
192.168.1.4
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
acl "domains". Falling back to Digest auth.
2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
v=0
o=MxSIP 0 1 IN IP4 192.168.1.4
s=SIP Call
c=IN IP4 192.168.1.4
t=0 0
m=audio 16384 RTP/AVP 0 110 8 9
a=rtpmap:0 PCMU/8000
a=rtpmap:110 PCMU/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=silenceSupp:off - - - -
a=ptime:20

2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
(sofia/internal/102 at 192.168.1.6:5060) State INIT
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85
sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
(sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
(sofia/internal/102 at 192.168.1.6:5060) State ROUTING
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148
sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
<102>->2001 in context default
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call_direction] continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_direction]
${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
set(call_direction=local)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tod_example]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
[tod_example] break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->holiday_example] continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
[holiday_example] break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [group-intercept]
destination_number(2001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe
r})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
destination_number(2001) =~ /^\*9001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
destination_number(2001) =~ /^\*9000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call_privacy] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call-group-simo]
destination_number(2001) =~ /^\*\*82(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [intercept-ext]
destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->send_to_voicemail] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.10d]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
destination_number(2001) =~ /^9(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.11d]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
destination_number(2001) =~ /^9(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->SipBroker.SIPDial] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->SipBroker.Toll_Free] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[SipBroker.Toll_Free] destination_number(2001) =~
/^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->TollFreeGateway] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [TollFreeGateway]
destination_number(2001) =~ /^91?(8(00|55|66|77|88)[2-9]\d{6})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->FlowRoute.91d10or11] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7519.9.10d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
caller_id_number(102) =~ /519/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7519.9.11d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
caller_id_number(102) =~ /519/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7612.9.10d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
${toll_allow}(local,domestic) =~ /local/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
destination_number(2001) =~ /^9(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7612.9.11d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
${toll_allow}(local,domestic) =~ /local/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
destination_number(2001) =~ /^9(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
destination_number(2001) =~ /^7(\d{7})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d10]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
destination_number(2001) =~ /^8(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d11]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
destination_number(2001) =~ /^8(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Time] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Time] destination_number(2001) =~ /^9170$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Date] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Date] destination_number(2001) =~ /^9171$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Date and Time] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Recordings]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
destination_number(2001) =~ /^\*(732)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Company_Directory] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Hersch_Lauren_Main] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Holiday_Announcement] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
admin phones.park] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
admin phones] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
phones] destination_number(2001) =~ /^1001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
destination_number(2001) =~ /^operator$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
destination_number(2001) =~ /^vmain$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
destination_number(2001) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [nb_conferences]
destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wb_conferences]
destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [uwb_conferences]
destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(2001) =~
/^\*9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
destination_number(2001) =~ /^\*5000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->parking_slots] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [parking_slots]
destination_number(2001) =~ /^(\*59[0-9][0-9])$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->valet_park_in] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_in]
destination_number(2001) =~ /^\*(6000)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->valet_park_out] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_out]
destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
destination_number(2001) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
destination_number(2001) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
destination_number(2001) =~ /callpark/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
destination_number(2001) =~ /pickup/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
destination_number(2001) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->fax_receive]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
destination_number(2001) =~ /^\*9178$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->fax_transmit] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
destination_number(2001) =~ /^\*9179$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_180] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
destination_number(2001) =~ /^\*9180$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
destination_number(2001) =~ /^\*9191$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->show_info]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
destination_number(2001) =~ /^\*9192$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->video_record] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
destination_number(2001) =~ /^\*9193$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->video_playback] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_playback]
destination_number(2001) =~ /^\*9194$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->delay_echo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
destination_number(2001) =~ /^\*9195$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
destination_number(2001) =~ /^\*9196$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->milliwatt]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
destination_number(2001) =~ /^\*9197$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tone_stream]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
destination_number(2001) =~ /^\*9198$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->hold_music]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
destination_number(2001) =~ /^\*9664$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [Local_Extension]
destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(dialed_extension=2001)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
export(dialed_extension=2001)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db ${domain_name}
2001 ${limit_max} ${limit_destination})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s
trftime(%b)}/${strftime(%d)}/${uuid}.wav)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
execute_extension::att_xfer XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
4000, 440.0, 480.0))
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(continue_on_fail=true)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe
r})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
bridge(user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
${domain_name} ${dialed_extension})
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> CS_EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
(sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241
sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial/102/2001)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c
1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
2012 23:50:41 -0400)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012
23:50:41 -0400]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
(export_vars) [dialed_extension]=[2001]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
192.168.1.6_2001 max:5, interval:0
2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001 is
now 1/5
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
execute_extension::dx XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
execute_extension::dx XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bd
e-116c-4100-b269-a6be7455c6c1.wav)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bd
e-116c-4100-b269-a6be7455c6c1.wav
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
execute_extension::cf XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
execute_extension::cf XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
execute_extension::att_xfer XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
execute_extension::att_xfer XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
440.0, 480.0))
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0,
480.0)]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
4000, 440.0, 480.0))
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000,
440.0, 480.0)]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-call_return/2001/102)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c
6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
[dialed_extension]=[2001] to event
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
[dialed_extension]=[2001] to event
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
[sip_invite_domain]=[192.168.1.6]
2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
[presence_id]=[2001 at 192.168.1.6]
2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111
sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
Disable 2833 dtmf and switch to INFO
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
[sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 192.168.1.4
port 16384 codec: 0 ms: 20
2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms
2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
sofia/internal/102 at 192.168.1.6:5060!
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
sofia/internal/102 at 192.168.1.6:5060:
v=0
o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
s=FreeSWITCH
c=IN IP4 192.168.1.6
t=0 0
m=audio 23164 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
[sofia/internal/102 at 192.168.1.6:5060] has been answered
2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
confirmed.
EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
2001)
2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
Activated L16 at 8000hz 1 channels 20ms
2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup
sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
sofia/internal/102 at 192.168.1.6:5060 [KILL]
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
file
/usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.
wav
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
sofia/internal/102 at 192.168.1.6:5060 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/102 at 192.168.1.6:5060) State HANGUP
2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel
sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: NORMAL_CLEARING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> CS_REPORTING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/102 at 192.168.1.6:5060) State REPORTING
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
NORMAL_CLEARING
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
CS_DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
(sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
(sofia/internal/102 at 192.168.1.6:5060) Ended
2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close Channel
sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/102 at 192.168.1.6:5060) State DESTROY
2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374
sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
freeswitch at internal> /bye
[root at server bin]#

Regards,

Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.com

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From anton.jugatsu at gmail.com  Mon Apr  2 10:18:55 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Mon, 2 Apr 2012 10:18:55 +0400
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <000301cd108a$4d3f55a0$e7be00e0$@com>
References: <000301cd108a$4d3f55a0$e7be00e0$@com>
Message-ID: <CANBhyPBtJC5uBxZnC=YtWCj5AFx-g9imD=O3WzKiUQZ5sDQEBw@mail.gmail.com>

I think that the best way to check endpoints registrations is sofia status
profile internal reg. I have sometimes accuracy problems with show
registrations.

2 ?????? 2012 ?. 8:37 ???????????? Phil Quesinberry <
philq at qsystemsengineering.com> ???????:

> **
>
> Hi,
>
> I've noticed what appears to be a bug which I wanted to bring to your
> attention.  Sometimes when FS is started, extensions can't be called even
> after they've registered.  This doesn't happen all the time... and
> restarting FS always solves the problem.  Just to be clear - this problem only
> sometimes occurs when FS has been started/restarted, like after an
> update.  Once FS has "started successfully" and sees presence for
> registered extensions, it continues to work fine.
>
> It's all or nothing - no extensions can be dialed when this issue is
> present, whether they're local or NATted.  If one extension can be
> reached, you can reach them all (if they're registered, of course).
>
> See debug info below on an attempt to dial a local (no NAT) extension -this was one of the recent beta git pulls and the problem has been present
> through several updates.  Let me know if you need any additional info.
>
> freeswitch at internal> show registrations
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>
> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060
> ;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>
> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.182:5060
> ;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>
> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060
> ;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>
> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.222.182:1066
> ;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>
> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.222.182:1068
> ;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>
> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.222.182:1067
> ;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>
>
> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.100.94:54629
> ;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>
>
> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.100.94:6480
> ;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>
> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/
> sip:519 at 74.93.222.182:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>
> 9 total.
>
> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
> 192.168.1.4
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [received][100]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>
> v=0
>
> o=MxSIP 0 1 IN IP4 192.168.1.4
>
> s=SIP Call
>
> c=IN IP4 192.168.1.4
>
> t=0 0
>
> m=audio 16384 RTP/AVP 0 110 8 9
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:110 PCMU/16000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:9 G722/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746 (sofia/internal/
> 102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125 (sofia/internal/
> 102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>
> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
> <102>->2001 in context default
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_direction] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
> set(call_direction=local)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->tod_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [tod_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->holiday_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [holiday_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group-intercept] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group-intercept] destination_number(2001) =~ /^\*8$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->snom-demo-2] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
> destination_number(2001) =~ /^\*9001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->snom-demo-1] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
> destination_number(2001) =~ /^\*9000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_privacy] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
> destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_return] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
> destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->del-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->add-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-simo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call-group-simo] destination_number(2001) =~ /^\*\*82(\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-order] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->intercept-ext] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [intercept-ext] destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension_Skinny] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->send_to_voicemail] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->enum.9.10d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->enum.9.11d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.SIPDial] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.Toll_Free] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.Toll_Free] destination_number(2001) =~
> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->TollFreeGateway] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [TollFreeGateway] destination_number(2001) =~
> /^91?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->FlowRoute.91d10or11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->xmpp.8d10] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->xmpp.8d11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Time] destination_number(2001) =~ /^9170$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date] destination_number(2001) =~ /^9171$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date and Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Recordings] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
> destination_number(2001) =~ /^\*(732)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Company_Directory] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Hersch_Lauren_Main] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Holiday_Announcement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones.park] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all
> admin phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all
> admin phones] destination_number(2001) =~ /^1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_sales] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_support] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_billing] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
> destination_number(2001) =~ /^operator$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
> destination_number(2001) =~ /^vmain$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->nb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [nb_conferences] destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->wb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [wb_conferences] destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->uwb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [uwb_conferences] destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->cdquality_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(2001) =~
> /^\*9(888|8888|1616|3232)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
> destination_number(2001) =~ /^\*5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->dynamic_conference] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->rtp_multicast_page] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->parking_slots] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [parking_slots] destination_number(2001) =~ /^(\*59[0-9][0-9])$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_in] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [valet_park_in] destination_number(2001) =~ /^\*(6000)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_out] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [valet_park_out] destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /^parking$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /callpark/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /pickup/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
> destination_number(2001) =~ /^wait$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->fax_receive] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
> destination_number(2001) =~ /^\*9178$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->fax_transmit] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
> destination_number(2001) =~ /^\*9179$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_180] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
> destination_number(2001) =~ /^\*9180$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_music_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
> destination_number(2001) =~ /^\*9191$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->show_info] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
> destination_number(2001) =~ /^\*9192$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_record] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
> destination_number(2001) =~ /^\*9193$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_playback] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->delay_echo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
> destination_number(2001) =~ /^\*9195$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
> destination_number(2001) =~ /^\*9196$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->milliwatt] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
> destination_number(2001) =~ /^\*9197$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->tone_stream] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
> destination_number(2001) =~ /^\*9198$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->zrtp_enrollement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->hold_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
> destination_number(2001) =~ /^\*9664$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS)
> [Local_Extension] destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> export(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db
> ${domain_name} 2001 ${limit_max} ${limit_destination})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
> 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(hangup_after_bridge=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(continue_on_fail=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> bridge(user/${user_data(${destination_number}@${domain_name} attr
> id)}@${domain_name})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
> ${domain_name} ${dialed_extension})
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [call_direction]=[local]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial/102/2001)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
> 2012 23:50:41 -0400)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012 23:50:41 -0400]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
> (export_vars) [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>
> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
> 192.168.1.6_2001 max:5, interval:0
>
> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001
> is now 1/5
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
> execute_extension::dx XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
> execute_extension::cf XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
> execute_extension::att_xfer XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
> 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
> 4000, 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000, 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-call_return/2001/102)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 sofia/internal/
> 102 at 192.168.1.6:5060 EXPORTING[export_vars] [dialed_extension]=[2001] to
> event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 sofia/internal/
> 102 at 192.168.1.6:5060 EXPORTING[export_vars] [dialed_extension]=[2001] to
> event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [sip_invite_domain]=[192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [presence_id]=[2001 at 192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
> Cause: USER_NOT_REGISTERED
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111 sofia/internal/
> 102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
> Disable 2833 dtmf and switch to INFO
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 ->
> 192.168.1.4 port 16384 codec: 0 ms: 20
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
>
> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
> sofia/internal/102 at 192.168.1.6:5060!
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
> sofia/internal/102 at 192.168.1.6:5060:
>
> v=0
>
> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.1.6
>
> t=0 0
>
> m=audio 23164 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>
> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [completed][200]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [ready][200]
>
> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
> confirmed.
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
> 2001)
>
> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>
> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup sofia/internal/
> 102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
> file /usr/local/freeswitch/storage/voicemail/default/
> 192.168.1.6/2001/greeting_1.wav
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
> sofia/internal/102 at 192.168.1.6:5060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel sofia/internal/
> 102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
> CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Ended
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close
> Channel sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>
> freeswitch at internal> /bye
>
> [root at server bin]#
>
> Regards,
>
> *******Phil Quesinberry*
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.com>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From brian at freeswitch.org  Mon Apr  2 10:26:00 2012
From: brian at freeswitch.org (Brian West)
Date: Mon, 2 Apr 2012 01:26:00 -0500
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <CANBhyPBtJC5uBxZnC=YtWCj5AFx-g9imD=O3WzKiUQZ5sDQEBw@mail.gmail.com>
References: <000301cd108a$4d3f55a0$e7be00e0$@com>
	<CANBhyPBtJC5uBxZnC=YtWCj5AFx-g9imD=O3WzKiUQZ5sDQEBw@mail.gmail.com>
Message-ID: <-82712645424086336@unknownmsgid>

And use Jira to report bugs please.

/b

Sent from my iPhone

On Apr 2, 2012, at 1:20 AM, Anton Kvashenkin <anton.jugatsu at gmail.com>
wrote:

I think that the best way to check endpoints registrations is sofia status
profile internal reg. I have sometimes accuracy problems with show
registrations.

2 ?????? 2012 ?. 8:37 ???????????? Phil Quesinberry <
philq at qsystemsengineering.com> ???????:

> **
>
> Hi,
>
> I?ve noticed what appears to be a bug which I wanted to bring to your
> attention.  Sometimes when FS is started, extensions can?t be called even
> after they?ve registered.  This doesn?t happen all the time? and
> restarting FS always solves the problem.  Just to be clear - this problem only
> sometimes occurs when FS has been started/restarted, like after an
> update.  Once FS has ?started successfully? and sees presence for
> registered extensions, it continues to work fine.
>
> It?s all or nothing ? no extensions can be dialed when this issue is
> present, whether they?re local or NATted.  If one extension can be
> reached, you can reach them all (if they?re registered, of course).
>
> See debug info below on an attempt to dial a local (no NAT) extension ?this was one of the recent beta git pulls and the problem has been present
> through several updates.  Let me know if you need any additional info.
>
> freeswitch at internal> show registrations
>
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>
> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060
> ;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>
> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.182:5060
> ;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>
> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060
> ;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>
> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.222.182:1066
> ;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>
> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.222.182:1068
> ;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>
> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.222.182:1067
> ;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>
>
> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.100.94:54629
> ;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>
>
> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.100.94:6480
> ;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>
> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/
> sip:519 at 74.93.222.182:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>
> 9 total.
>
> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
> 192.168.1.4
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [received][100]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>
> v=0
>
> o=MxSIP 0 1 IN IP4 192.168.1.4
>
> s=SIP Call
>
> c=IN IP4 192.168.1.4
>
> t=0 0
>
> m=audio 16384 RTP/AVP 0 110 8 9
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:110 PCMU/16000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:9 G722/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746 (sofia/internal/
> 102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125 (sofia/internal/
> 102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>
> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
> <102>->2001 in context default
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_direction] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
> set(call_direction=local)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->tod_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [tod_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->holiday_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [holiday_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group-intercept] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group-intercept] destination_number(2001) =~ /^\*8$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${sip_has_crypto}() =~
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->snom-demo-2] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
> destination_number(2001) =~ /^\*9001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->snom-demo-1] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
> destination_number(2001) =~ /^\*9000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_privacy] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
> destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_return] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
> destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->del-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->add-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-simo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call-group-simo] destination_number(2001) =~ /^\*\*82(\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-order] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->intercept-ext] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [intercept-ext] destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension_Skinny] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->send_to_voicemail] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->enum.9.10d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->enum.9.11d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.SIPDial] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.Toll_Free] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.Toll_Free] destination_number(2001) =~
> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->TollFreeGateway] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [TollFreeGateway] destination_number(2001) =~
> /^91?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->FlowRoute.91d10or11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->xmpp.8d10] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->xmpp.8d11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Time] destination_number(2001) =~ /^9170$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date] destination_number(2001) =~ /^9171$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date and Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Recordings] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
> destination_number(2001) =~ /^\*(732)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Company_Directory] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Hersch_Lauren_Main] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Holiday_Announcement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones.park] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all
> admin phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all
> admin phones] destination_number(2001) =~ /^1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_sales] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_support] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_billing] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
> destination_number(2001) =~ /^operator$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
> destination_number(2001) =~ /^vmain$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->nb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [nb_conferences] destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->wb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [wb_conferences] destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->uwb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [uwb_conferences] destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->cdquality_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(2001) =~
> /^\*9(888|8888|1616|3232)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
> destination_number(2001) =~ /^\*5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->dynamic_conference] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->rtp_multicast_page] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->parking_slots] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [parking_slots] destination_number(2001) =~ /^(\*59[0-9][0-9])$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_in] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [valet_park_in] destination_number(2001) =~ /^\*(6000)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_out] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [valet_park_out] destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /^parking$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /callpark/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /pickup/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
> destination_number(2001) =~ /^wait$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->fax_receive] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
> destination_number(2001) =~ /^\*9178$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->fax_transmit] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
> destination_number(2001) =~ /^\*9179$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_180] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
> destination_number(2001) =~ /^\*9180$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_music_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
> destination_number(2001) =~ /^\*9191$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->show_info] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
> destination_number(2001) =~ /^\*9192$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_record] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
> destination_number(2001) =~ /^\*9193$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_playback] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->delay_echo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
> destination_number(2001) =~ /^\*9195$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
> destination_number(2001) =~ /^\*9196$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->milliwatt] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
> destination_number(2001) =~ /^\*9197$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->tone_stream] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
> destination_number(2001) =~ /^\*9198$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->zrtp_enrollement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->hold_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
> destination_number(2001) =~ /^\*9664$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS)
> [Local_Extension] destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> export(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db
> ${domain_name} 2001 ${limit_max} ${limit_destination})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
> 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(hangup_after_bridge=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(continue_on_fail=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> bridge(user/${user_data(${destination_number}@${domain_name} attr
> id)}@${domain_name})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
> ${domain_name} ${dialed_extension})
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [call_direction]=[local]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial/102/2001)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
> 2012 23:50:41 -0400)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012 23:50:41 -0400]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
> (export_vars) [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>
> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
> 192.168.1.6_2001 max:5, interval:0
>
> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001
> is now 1/5
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
> execute_extension::dx XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
> execute_extension::cf XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
> execute_extension::att_xfer XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
> 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
> 4000, 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000, 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-call_return/2001/102)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 sofia/internal/
> 102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 sofia/internal/
> 102 at 192.168.1.6:5060 EXPORTING[export_vars] [dialed_extension]=[2001] to
> event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 sofia/internal/
> 102 at 192.168.1.6:5060 EXPORTING[export_vars] [dialed_extension]=[2001] to
> event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [sip_invite_domain]=[192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [presence_id]=[2001 at 192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
> Cause: USER_NOT_REGISTERED
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111 sofia/internal/
> 102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
> Disable 2833 dtmf and switch to INFO
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 ->
> 192.168.1.4 port 16384 codec: 0 ms: 20
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
>
> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
> sofia/internal/102 at 192.168.1.6:5060!
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
> sofia/internal/102 at 192.168.1.6:5060:
>
> v=0
>
> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.1.6
>
> t=0 0
>
> m=audio 23164 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>
> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [completed][200]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel sofia/internal/
> 102 at 192.168.1.6:5060 entering state [ready][200]
>
> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
> confirmed.
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
> 2001)
>
> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848 (sofia/internal/
> 102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>
> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup sofia/internal/
> 102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
> file /usr/local/freeswitch/storage/voicemail/default/
> 192.168.1.6/2001/greeting_1.wav
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
> sofia/internal/102 at 192.168.1.6:5060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel sofia/internal/
> 102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause:
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
> CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Ended
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close
> Channel sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374 sofia/internal/
> 102 at 192.168.1.6:5060 SOFIA DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>
> freeswitch at internal> /bye
>
> [root at server bin]#
>
> Regards,
>
> *******Phil Quesinberry*
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> *****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.com>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From Daniel.Knaggs at realitysolutions.co.uk  Mon Apr  2 12:16:56 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Mon, 2 Apr 2012 08:16:56 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>

I appreciate your honesty, Gabe.  But seriously, is that the reason no-one else is responding or is it because no-one has any ideas?

Unfortunately, I don?t (and can?t) control what appears on my signature as it?s added by our e-mail gateway not my e-mail client (if it can?t see a signature on the e-mail it will add it).

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Gunderson
Sent: 31 March 2012 09:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Any ideas, anyone?

I'll be honest. I haven't followed the thread because of the loud, in-your-face advertisement in the email footer. Nobody is going to make you remove it, but they might respond to more of your emails if you show them that you respect the list by not cluttering it with noise.

Just my 2 cents.  And of course good luck on the issue.


Gabe

[cid:image92a6e6.PNG at a012bb36.4aabcf94]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:image3dcd97.PNG at 2cd2cd92.4fa6ec40]





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From anita.hall at simmortel.com  Mon Apr  2 14:18:18 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Mon, 2 Apr 2012 15:48:18 +0530
Subject: [Freeswitch-users] fax tone testing
Message-ID: <CAGKPtBuuxv3Vued4FTySPpiYcQ_57fO5Z5CjW3z3cwCHhY93aA@mail.gmail.com>

Hi

I suspect that my telco set-up is acting funny and I want to use spectral
analysis to confirm the culprit :)

What is the best way to generate Fax tones from a dialplan and then record
them at the other end? Also, where can I get a list of the all the tones
and duration which are used in Fax.


Thanks.

regards,
Anita
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From sylver_b at yahoo.com  Mon Apr  2 15:34:07 2012
From: sylver_b at yahoo.com (sylver_b)
Date: Mon, 2 Apr 2012 04:34:07 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1333366447.52788.YahooMailMobile@web162802.mail.bf1.yahoo.com>

<a href="http://c1.00364.com/blog.old/wp-content/plugins/extended-comment-options/02efpk.html"> http://c1.00364.com/blog.old/wp-content/plugins/extended-comment-options/02efpk.html</a>
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From wesleyakio at tuntscorp.com  Mon Apr  2 15:56:43 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Mon, 2 Apr 2012 08:56:43 -0300
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
Message-ID: <CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>

If all you wish is to purge the tones from your stream you can simply
capture them in the dialplan, no need to mess with the source...

Best,

Wesley Akio
Em 02/04/2012 05:17, "Daniel Knaggs" <Daniel.Knaggs at realitysolutions.co.uk>
escreveu:

>   I appreciate your honesty, Gabe.  But seriously, is that the reason
> no-one else is responding or is it because no-one has any ideas?****
>
> ** **
>
> Unfortunately, I don?t (and can?t) control what appears on my signature as
> it?s added by our e-mail gateway not my e-mail client (if it can?t see a
> signature on the e-mail it will add it).****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gabriel
> Gunderson
> *Sent:* 31 March 2012 09:12
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls****
>
> ** **
>
> On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <
> Daniel.Knaggs at realitysolutions.co.uk> wrote:****
>
> Any ideas, anyone?****
>
>  ** **
>
> I'll be honest. I haven't followed the thread because of the loud,
> in-your-face advertisement in the email footer. Nobody is going to make you
> remove it, but they might respond to more of your emails if you show them
> that you respect the list by not cluttering it with noise.****
>
> ** **
>
> Just my 2 cents.  And of course good luck on the issue.****
>
> ** **
>
> ** **
>
> Gabe****
>
>      Daniel Knaggs
>
> Software Developer
>     Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston
> upon Hull, East Yorkshire, HU7 0AE
> Tel: 01482 828000 / Fax: 01482 373100
> Daniel.Knaggs at realitysolutions.co.uk
> www.realitysolutions.co.uk
> ------------------------------
>   Sage Accredited Business Partner serving businesses in Yorkshire &
> Lincolnshire
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From miha at softnet.si  Mon Apr  2 16:25:31 2012
From: miha at softnet.si (Miha)
Date: Mon, 02 Apr 2012 14:25:31 +0200
Subject: [Freeswitch-users] mod_cdr_cvs
Message-ID: <4F799ABB.2050505@softnet.si>

Hi,

as I needed quick solution to put cdr to mysql I have used perl script 
which is at the and of wiki page of mod_cdr_cvs.

As I have now 500 registrations on FS there are a lot of cdr files 
created in log/cdr-cvs/. I will need a little more time to create my 
script and parser for xml_cdr so I have one question.

Can I delete all thous cdr-cvs files? Will FS created Master file when 
the first call will be made after I will deleted all cdr-cvs files?

Thank you!

Regards,
Miha


From Daniel.Knaggs at realitysolutions.co.uk  Mon Apr  2 16:29:52 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Mon, 2 Apr 2012 12:29:52 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>

Interesting, how would I do this?  Can't use "bind_meta_app" since that only works with numbers...  The tones appear from the caller not the callee (leg A).

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio
Sent: 02 April 2012 12:57
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls


If all you wish is to purge the tones from your stream you can simply capture them in the dialplan, no need to mess with the source...

Best,

Wesley Akio
Em 02/04/2012 05:17, "Daniel Knaggs" <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> escreveu:
I appreciate your honesty, Gabe.  But seriously, is that the reason no-one else is responding or is it because no-one has any ideas?

Unfortunately, I don't (and can't) control what appears on my signature as it's added by our e-mail gateway not my e-mail client (if it can't see a signature on the e-mail it will add it).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Gabriel Gunderson
Sent: 31 March 2012 09:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Any ideas, anyone?

I'll be honest. I haven't followed the thread because of the loud, in-your-face advertisement in the email footer. Nobody is going to make you remove it, but they might respond to more of your emails if you show them that you respect the list by not cluttering it with noise.

Just my 2 cents.  And of course good luck on the issue.


Gabe
[cid:image001.png at 01CD10D4.AF0ABC80]


Daniel Knaggs

Software Developer




Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>

________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire



[cid:image002.png at 01CD10D4.AF0ABC80]



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sardor.tukhtaev at gmail.com  Mon Apr  2 11:07:20 2012
From: sardor.tukhtaev at gmail.com (Sardor Tukhtaev)
Date: Mon, 2 Apr 2012 12:07:20 +0500
Subject: [Freeswitch-users] problems LuaSql in FreeSwitch
In-Reply-To: <5808463933582034130@unknownmsgid>
References: <CAPu2oO+2wX_Y2XhP-xbMnrBtqLVsopqXt=qoFHhMVhqq6TNWew@mail.gmail.com>
	<5808463933582034130@unknownmsgid>
Message-ID: <CAPu2oOKkV9eX_hfyo7vFqc6vy_JwPfXqo_FA-+Q_Mr4MJeWUKw@mail.gmail.com>

Hello everybody,

The problem arises from connecting to the database (Oracle). In the console
[fs_cli]
this error: >>>
2012-03-28 17:21:57.558786 [ERR] mod_lua.cpp: 198 C: \ Program Files
\ FreeSWITCH \ scr
ipts / db_connect.lua: 4: module 'luasql.oci8' not found:
         no field package.preload ['luasql.oci8']
         no file '. \ luasql \ oci8.lua'
         no file 'C: \ Program Files \ FreeSWITCH \ lua \ luasql \ oci8.lua'
         no file 'C: \ Program Files \ FreeSWITCH \ lua \ luasql \ oci8 \
init.lua'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8.lua'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8 \
init.lua'
         no file 'C: \ Program Files \ Lua \ 5.1 \ lua \ luasql \ oci8.luac'
         no file '. \ luasql \ oci8.dll'
         no file '. \ luasql \ oci851.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci851.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql \
oci8.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql \
oci851.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ loadall.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ loadall.dll'
         no file '. \ luasql.dll'
         no file '. \ luasql51.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ luasql51.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql51.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ loadall.dll'
         no file 'C: \ Program Files \ FreeSWITCH \ clibs \ loadall.dll'
stack traceback:
         [C]: in function 'require'
         C: \ Program Files \ FreeSWITCH \ scripts / db_connect.lua: 4: in
main chunk

These files I didn't find. Set version LuaForWindows_v5.1.4-45
There was written, that it supports Oracle. However,the files with the
extension[oci8] I haven't found.
Could you help me with this problem.
Thanks in advance.
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From alec.taylor6 at gmail.com  Mon Apr  2 19:13:38 2012
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Tue, 3 Apr 2012 01:13:38 +1000
Subject: [Freeswitch-users] problems LuaSql in FreeSwitch
In-Reply-To: <CAPu2oOKkV9eX_hfyo7vFqc6vy_JwPfXqo_FA-+Q_Mr4MJeWUKw@mail.gmail.com>
References: <CAPu2oO+2wX_Y2XhP-xbMnrBtqLVsopqXt=qoFHhMVhqq6TNWew@mail.gmail.com>
	<5808463933582034130@unknownmsgid>
	<CAPu2oOKkV9eX_hfyo7vFqc6vy_JwPfXqo_FA-+Q_Mr4MJeWUKw@mail.gmail.com>
Message-ID: <CAO+9iGfdiuos4YrpSRHR4oSauvukuMQMUbMHdkj-5+z1umZ7CQ@mail.gmail.com>

Try installing the extentions?

On Mon, Apr 2, 2012 at 5:07 PM, Sardor Tukhtaev
<sardor.tukhtaev at gmail.com> wrote:
>
>
> Hello everybody,
>
> The problem arises?from connecting?to the database?(Oracle).?In?the
> console?[fs_cli]
> this error: >>>
> 2012-03-28 17:21:57.558786 [ERR] mod_lua.cpp:?198?C: \ Program Files
> \ FreeSWITCH \ scr
> ipts / db_connect.lua:?4: module 'luasql.oci8' not found:
> ?????????no field package.preload ['luasql.oci8']
> ?????????no file '. \ luasql \ oci8.lua'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ lua \ luasql \ oci8.lua'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ lua \ luasql \ oci8 \
> init.lua'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8.lua'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8 \
> init.lua'
> ?????????no file 'C: \ Program Files \ Lua \ 5.1 \ lua \ luasql \ oci8.luac'
> ?????????no file '. \ luasql \ oci8.dll'
> ?????????no file '. \ luasql \ oci851.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci8.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql \ oci851.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql \
> oci8.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql \
> oci851.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ loadall.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ loadall.dll'
> ?????????no file '. \ luasql.dll'
> ?????????no file '. \ luasql51.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ luasql51.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ luasql51.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ loadall.dll'
> ?????????no file 'C: \ Program Files \ FreeSWITCH \ clibs \ loadall.dll'
> stack traceback:
> ?????????[C]: in function 'require'
> ?????????C: \ Program Files \ FreeSWITCH \ scripts / db_connect.lua:?4: in
> main chunk
>
> These files?I didn't find.?Set?version?LuaForWindows_v5.1.4-45
> There was written, that it?supports?Oracle.?However,the?files with the
> extension[oci8] I haven't found.
> Could you help me with this problem.
> Thanks in advance.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Mon Apr  2 19:43:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 08:43:48 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>

Maybe bind_digit_action would capture these for you. You could try binding
something simple, like a log function:

<action application="bind_digit_action" data="rogue_dtmf,~\w+,exec:log,INFO
Rogue digit(s): ${last_matching_digits}"/>
<action application="digit_action_set_realm" data="rogue_dtmf"/>

You'll need to experiment to make sure that regex works. If not you could
possibly do each digit explicity, like this:
<action application="bind_digit_action" data="rogue_dtmf,A,exec:log,INFO
Rogue digit: DTMF 'A' received"/>
<action application="bind_digit_action" data="rogue_dtmf,B,exec:log,INFO
Rogue digit: DTMF 'B' received"/>
<action application="bind_digit_action" data="rogue_dtmf,C,exec:log,INFO
Rogue digit: DTMF 'C' received"/>
<action application="bind_digit_action" data="rogue_dtmf,D,exec:log,INFO
Rogue digit: DTMF 'D' received"/>

I don't honestly know if this will work, but I'd like to see what happens
in your case. FYI, you can read up on bind_digit_action (AKA 'bda') here:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action

-MC

On Mon, Apr 2, 2012 at 5:29 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

>  Interesting, how would I do this?  Can?t use ?bind_meta_app? since that
> only works with numbers?  The tones appear from the caller not the callee
> (leg A).****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio
> *Sent:* 02 April 2012 12:57
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls****
>
> ** **
>
> If all you wish is to purge the tones from your stream you can simply
> capture them in the dialplan, no need to mess with the source...****
>
> Best,****
>
> Wesley Akio****
>
> Em 02/04/2012 05:17, "Daniel Knaggs" <Daniel.Knaggs at realitysolutions.co.uk>
> escreveu:****
>
> I appreciate your honesty, Gabe.  But seriously, is that the reason no-one
> else is responding or is it because no-one has any ideas?****
>
>  ****
>
> Unfortunately, I don?t (and can?t) control what appears on my signature as
> it?s added by our e-mail gateway not my e-mail client (if it can?t see a
> signature on the e-mail it will add it).****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gabriel
> Gunderson
> *Sent:* 31 March 2012 09:12
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls****
>
>  ****
>
> On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <
> Daniel.Knaggs at realitysolutions.co.uk> wrote:****
>
> Any ideas, anyone?****
>
>   ****
>
> I'll be honest. I haven't followed the thread because of the loud,
> in-your-face advertisement in the email footer. Nobody is going to make you
> remove it, but they might respond to more of your emails if you show them
> that you respect the list by not cluttering it with noise.****
>
>  ****
>
> Just my 2 cents.  And of course good luck on the issue.****
>
>  ****
>
>
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From fs-list at communicatefreely.net  Mon Apr  2 19:49:39 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Mon, 02 Apr 2012 11:49:39 -0400
Subject: [Freeswitch-users] Session-wise Language Setting
In-Reply-To: <13D710C546A14C09801E1CC46A394BB4@sharad>
References: <mailman.6169.1332986617.13443.freeswitch-users@lists.freeswitch.org><4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad><4F746EFF.3050305@communicatefreely.net><00C2DD790B254A1BACD50CAA66B123B9@sharad><1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se><12BA6FAF92674DEE9650CF4829B95DF0@sharad>	<CAKzWOxWus_Ryo97ARrJYbFjaJrmPEUxDF169iqzLKbw7AaW+DQ@mail.gmail.com>
	<13D710C546A14C09801E1CC46A394BB4@sharad>
Message-ID: <4F79CA93.3030205@communicatefreely.net>

Hi Sharad,

When you specify a sound file to play like that, it uses a prefix 
defined (usually) in vars.xml, and it doesn't expand to the language.  
The assumption by the designers I assume was that you would change the 
prefix to your language.

For multi-language systems, you should always use phrase macros.

On our system, we made several custom phrase macros that are very 
simple, but will select the correct audio file based on the language.

Any time you would normally specify an audio file, you can instead use 
phrase:my_macro:data

Then, create a phrase macro for each language that will match your data 
and play the appropriate audio file.

It could be something as simple as this:
English:

<macro name="ivr-file">
      <input pattern="^(\w+)$">
        <match>
            <action function="play-file" 
data="$${media_dir}/en/us/callie/ivr/$1wav"/>
        </match>
      </input>
</macro

And then in your Russian phrase macros:

<macro name="ivr-file">
      <input pattern="^(\w+)$">
        <match>
            <action function="play-file" 
data="$${media_dir}/ru/RU/elena/ivr/$1wav"/>
        </match>
      </input>
</macro

Be creative here.  Phrase macros are your friend.  I would limit playing 
audio files to things like sound effects, or other non-verbal recordings.


Sharad Garg wrote:
>
> Hello Mr. MC
>
> Thanks for the link.....seems working.
>
> But observed another point i.e. it works only for the speech phrase 
> macros.
>
> When I write a simple xml dialplan like this -
>
>     <extension name="lang_test">
>          <condition field="destination_number" expression="^(3333)$">
>               <action application="set" data="bypass_media=true"/>
>               <action application="answer"/>
>               <action application="sleep" data="1000"/>
>               <action application="set" data="default_language=ru"/>
>               <action application="playback" data="ivr/test.wav"/>
>          </condition>
>     </extension>
>
> This dialplan still plays the test.wav which is in 
> <base_dir>/sounds/en/us/callie/ivr/8000/test.wav. While it should play 
> the test.wav from <base_dir>/sounds/ru/RU/elena/ivr/8000/test.wav. & 
> at the same time, speech phrase macro use the 'ru' language.
>
> So when I did my googling, I found that the language settings is only 
> for speech phrase macros. It is not implemented in Freeswitch to use 
> the desired languages at all places.
>
> So need to know is there any latest development on this ?
>
> Thanks in advance.
>
> Regards
>
> Sharad
>
>  
>
>     ----- Original Message -----
>     *From:* Michael Collins <mailto:msc at freeswitch.org>
>     *To:* FreeSWITCH Users Help
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Saturday, March 31, 2012 1:15 AM
>     *Subject:* Re: [Freeswitch-users] Session-wise Language Setting
>
>
>
>     On Fri, Mar 30, 2012 at 5:13 AM, Sharad Garg <sharad at coraltele.com
>     <mailto:sharad at coraltele.com>> wrote:
>
>         Thanks Peter...yes it works.
>
>         Just a last query - Is there any API for setting the same so
>         that it can be
>         used from Javascript.
>
>         Best Regards
>         Sharad
>
>
>     http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From muelbuesch at as-infodienste.de  Mon Apr  2 19:50:41 2012
From: muelbuesch at as-infodienste.de (=?ISO-8859-15?Q?Marcus_M=FClb=FCsch?=)
Date: Mon, 02 Apr 2012 17:50:41 +0200
Subject: [Freeswitch-users] Bridging connection analog
 line->freeswitch->outbound and getting meaningful data
Message-ID: <4F79CAD1.7080009@as-infodienste.de>

Hello all,

    before I ask my question: We have an old "Sangoma Technologies Corp. 
A104u Quad T1/E1 AFT" card. Getting a newer card for this server is out 
of question. To make it run we had to use "WANPIPE Release: 3.3.14.11" 
and could get it to work only with "FreeSWITCH Version 1.0.4 (exported)" 
and thus use openzap instead of freetdm. Duh. Now, there also is an 
"Digium, Inc. Wildcard TDM2400P 24-port analog card" in the server.

    Everything is up and running (finally), and we can successfully 
accept calls on the analog card, and freeswitch switches it through the 
Digital or VoIP lines outbound.

    However, there is the matter that we do not get meaningful (for us) 
billing data in the CDR:

    "${start_stamp}" and "${end_stamp}" are correct, but the 
"${answer_stamp}" shows the time where the incoming call is accepted by 
freeswitch. Since the call is accepted we also never get another 
"${hangup_cause}" than "NORMAL_CLEARING".

    However, we would like to have the time (and hangup_cause) where the 
*outbound* connection is answered. This is especially annoying if the 
outbound connection is refused; for example because the line is busy.

    How is it possible to bring freeswitch to use the outgoing 
connection to record the answer_stamp?

    Am I doing something completely wrong?

    Any help or pointers are thankfully accepeted!

Marcus



From =?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?=  Mon Apr  2 20:05:55 2012
From: =?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?= (=?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?=)
Date: Mon, 02 Apr 2012 18:05:55 +0200
Subject: [Freeswitch-users] Bridging connection analog
 line->freeswitch->outbound and getting meaningful data
In-Reply-To: <4F79CAD1.7080009@as-infodienste.de>
References: <4F79CAD1.7080009@as-infodienste.de>
Message-ID: <4F79CE63.7000700@as-infodienste.de>

Am 02.04.2012 17:50, schrieb Marcus M?lb?sch:

>      However, there is the matter that we do not get meaningful (for us)
> billing data in the CDR:
>
>      "${start_stamp}" and "${end_stamp}" are correct, but the
> "${answer_stamp}" shows the time where the incoming call is accepted by
> freeswitch. Since the call is accepted we also never get another
> "${hangup_cause}" than "NORMAL_CLEARING".

Duh!

    Another test (that I really should have done beforehand; but I 
thought I already did) showed that freeswitch starts playing a "hold 
music", so freeswitch actually does something that makes it start billing.

   However, I'm currently looking for how and where to disable that 
behaviour.

    Apologies for my crude testing, but I'm still not finished with my 
problem.

Marcus



From mario_fs at mgtech.com  Mon Apr  2 20:14:50 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 09:14:50 -0700
Subject: [Freeswitch-users] Internal profile not started at FS startup if
	extension uses TCP
Message-ID: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>

I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next? 
Mario G

From anthony.minessale at gmail.com  Mon Apr  2 20:35:49 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 11:35:49 -0500
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <-82712645424086336@unknownmsgid>
References: <000301cd108a$4d3f55a0$e7be00e0$@com>
	<CANBhyPBtJC5uBxZnC=YtWCj5AFx-g9imD=O3WzKiUQZ5sDQEBw@mail.gmail.com>
	<-82712645424086336@unknownmsgid>
Message-ID: <CAKbxfG9P_B9BYzh9QCx+GUuv5uWw1pC_WeYcXW29Gr9Z0+S+xA@mail.gmail.com>

and use pastebin.freeswitch.org or attachments for logs imagine
reading this from your mobile phone.

On Mon, Apr 2, 2012 at 1:26 AM, Brian West <brian at freeswitch.org> wrote:
> And use Jira to report bugs please.
>
> /b
>
> Sent from my iPhone
>
> On Apr 2, 2012, at 1:20 AM, Anton Kvashenkin <anton.jugatsu at gmail.com>
> wrote:
>
> I think that the best way to check endpoints registrations is sofia status
> profile internal reg. I have sometimes accuracy problems with show
> registrations.
>
> 2 ?????? 2012 ?. 8:37 ???????????? Phil Quesinberry
> <philq at qsystemsengineering.com> ???????:
>>
>> Hi,
>>
>> I've noticed what appears to be a bug which I wanted to bring to your
>> attention.  Sometimes when FS is started, extensions can't be called even
>> after they've registered.  This doesn't happen all the time... and restarting
>> FS always solves the problem.  Just to be clear - this problem only
>> sometimes occurs when FS has been started/restarted, like after an update.
>> Once FS has "started successfully" and sees presence for registered
>> extensions, it continues to work fine.
>>
>> It's all or nothing - no extensions can be dialed when this issue is
>> present, whether they're local or NATted.  If one extension can be reached,
>> you can reach them all (if they're registered, of course).
>>
>> See debug info below on an attempt to dial a local (no NAT) extension -
>> this was one of the recent beta git pulls and the problem has been present
>> through several updates.  Let me know if you need any additional info.
>>
>>
>> freeswitch at internal> show registrations
>>
>>
>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>>
>>
>> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>>
>>
>> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.182:5060;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>>
>>
>> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>>
>>
>> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.222.182:1066;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>>
>>
>> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.222.182:1068;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>>
>>
>> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.222.182:1067;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>>
>>
>> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.100.94:54629;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>>
>>
>> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.100.94:6480;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>>
>>
>> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.222.182:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>>
>> 9 total.
>>
>> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
>> acl "domains". Falling back to Digest auth.
>>
>> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
>> (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
>> 192.168.1.4
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
>> acl "domains". Falling back to Digest auth.
>>
>> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
>> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel
>> sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>>
>> v=0
>>
>> o=MxSIP 0 1 IN IP4 192.168.1.4
>>
>> s=SIP Call
>>
>> c=IN IP4 192.168.1.4
>>
>> t=0 0
>>
>> m=audio 16384 RTP/AVP 0 110 8 9
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=rtpmap:110 PCMU/16000
>>
>> a=rtpmap:8 PCMA/8000
>>
>> a=rtpmap:9 G722/8000
>>
>> a=silenceSupp:off - - - -
>>
>> a=ptime:20
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746
>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
>> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85
>> sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125
>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
>> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886
>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
>> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148
>> sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
>> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>>
>> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
>> <102>->2001 in context default
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->call_direction] continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
>> set(call_direction=local)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->tod_example] continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
>> [tod_example] break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->holiday_example] continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
>> [holiday_example] break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->group-intercept] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [group-intercept] destination_number(2001) =~ /^\*8$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
>> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
>> continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
>> ${call_debug}(false) =~ /^true$/ break=never
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
>> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
>> break=never
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-last_dial/global/${uuid})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->snom-demo-2] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
>> destination_number(2001) =~ /^\*9001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->snom-demo-1] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
>> destination_number(2001) =~ /^\*9000$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->call_privacy] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
>> destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->call_return] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
>> destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->del-group]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
>> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->add-group]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
>> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->call-group-simo] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [call-group-simo] destination_number(2001) =~ /^\*\*82(\d{2})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->call-group-order] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->intercept-ext] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [intercept-ext]
>> destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Local_Extension_Skinny] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->send_to_voicemail] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->enum.9.10d] continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
>> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->enum.9.11d] continue=true
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
>> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->SipBroker.SIPDial] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->SipBroker.Toll_Free] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [SipBroker.Toll_Free] destination_number(2001) =~
>> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->TollFreeGateway] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [TollFreeGateway] destination_number(2001) =~
>> /^91?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->FlowRoute.91d10or11] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->HL7519.9.10d] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
>> caller_id_number(102) =~ /519/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->HL7519.9.11d] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
>> caller_id_number(102) =~ /519/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->HL7612.9.10d] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
>> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
>> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->HL7612.9.11d] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
>> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
>> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
>> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d10]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
>> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d11]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
>> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>> Clock Time] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>> Time] destination_number(2001) =~ /^9170$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>> Clock Date] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>> Date] destination_number(2001) =~ /^9171$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>> Clock Date and Time] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>> Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Recordings] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
>> destination_number(2001) =~ /^\*(732)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Company_Directory] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Hersch_Lauren_Main] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Holiday_Announcement] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
>> admin phones.park] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
>> phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
>> admin phones] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
>> phones] destination_number(2001) =~ /^1001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->group_dial_sales] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->group_dial_support] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->group_dial_billing] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
>> destination_number(2001) =~ /^operator$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
>> destination_number(2001) =~ /^vmain$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
>> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
>> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
>> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->nb_conferences] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [nb_conferences] destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->wb_conferences] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [wb_conferences] destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->uwb_conferences] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [uwb_conferences] destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->cdquality_conferences] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->freeswitch_public_conf_via_sip] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [freeswitch_public_conf_via_sip] destination_number(2001) =~
>> /^\*9(888|8888|1616|3232)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
>> destination_number(2001) =~ /^\*5000$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->dynamic_conference] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->rtp_multicast_page] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->parking_slots] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [parking_slots]
>> destination_number(2001) =~ /^(\*59[0-9][0-9])$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->valet_park_in] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_in]
>> destination_number(2001) =~ /^\*(6000)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->valet_park_out] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [valet_park_out] destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
>> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
>> destination_number(2001) =~ /^parking$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
>> destination_number(2001) =~ /callpark/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
>> destination_number(2001) =~ /pickup/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
>> destination_number(2001) =~ /^wait$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->fax_receive] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
>> destination_number(2001) =~ /^\*9178$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->fax_transmit] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
>> destination_number(2001) =~ /^\*9179$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->ringback_180] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
>> destination_number(2001) =~ /^\*9180$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->ringback_183_uk_ring] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->ringback_183_music_ring] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->ringback_post_answer_uk_ring] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->ringback_post_answer_music] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
>> break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
>> destination_number(2001) =~ /^\*9191$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->show_info]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
>> destination_number(2001) =~ /^\*9192$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->video_record] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
>> destination_number(2001) =~ /^\*9193$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->video_playback] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->delay_echo] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
>> destination_number(2001) =~ /^\*9195$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
>> destination_number(2001) =~ /^\*9196$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->milliwatt]
>> continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
>> destination_number(2001) =~ /^\*9197$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->tone_stream] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
>> destination_number(2001) =~ /^\*9198$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->zrtp_enrollement] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->hold_music] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
>> destination_number(2001) =~ /^\*9664$/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>> [default->Local_Extension] continue=false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS)
>> [Local_Extension] destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(dialed_extension=2001)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> export(dialed_extension=2001)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db
>> ${domain_name} 2001 ${limit_max} ${limit_destination})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
>> execute_extension::dx XML features)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
>> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
>> execute_extension::cf XML features)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
>> execute_extension::att_xfer XML features)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
>> 4000, 440.0, 480.0))
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(hangup_after_bridge=true)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(continue_on_fail=true)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>> var callgroup)})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>> bridge(user/${user_data(${destination_number}@${domain_name} attr
>> id)}@${domain_name})
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>>
>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
>> ${domain_name} ${dialed_extension})
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> CS_EXECUTE
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
>> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
>> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241
>> sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
>> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-last_dial/102/2001)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
>> 2012 23:50:41 -0400)
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012
>> 23:50:41 -0400]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>>
>> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
>> (export_vars) [dialed_extension]=[2001]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>>
>> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
>> 192.168.1.6_2001 max:5, interval:0
>>
>> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001
>> is now 1/5
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
>> execute_extension::dx XML features)
>>
>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
>> execute_extension::dx XML features
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
>> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>>
>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
>> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
>> execute_extension::cf XML features)
>>
>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
>> execute_extension::cf XML features
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
>> execute_extension::att_xfer XML features)
>>
>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
>> execute_extension::att_xfer XML features
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
>> 440.0, 480.0))
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0,
>> 480.0)]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
>> 4000, 440.0, 480.0))
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000,
>> 440.0, 480.0)]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-call_return/2001/102)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>> sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
>> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
>> [dialed_extension]=[2001] to event
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
>> global variables
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
>> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
>> [dialed_extension]=[2001] to event
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
>> global variables
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
>> [sip_invite_domain]=[192.168.1.6]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
>> [presence_id]=[2001 at 192.168.1.6]
>>
>> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
>> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>>
>> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
>> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>>
>> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
>> Cause: USER_NOT_REGISTERED
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111
>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
>> Disable 2833 dtmf and switch to INFO
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
>> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 192.168.1.4
>> port 16384 codec: 0 ms: 20
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
>> 160 bytes per 20ms
>>
>> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
>> sofia/internal/102 at 192.168.1.6:5060!
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932
>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
>> sofia/internal/102 at 192.168.1.6:5060:
>>
>> v=0
>>
>> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>>
>> s=FreeSWITCH
>>
>> c=IN IP4 192.168.1.6
>>
>> t=0 0
>>
>> m=audio 23164 RTP/AVP 0
>>
>> a=rtpmap:0 PCMU/8000
>>
>> a=silenceSupp:off - - - -
>>
>> a=ptime:20
>>
>> a=sendrecv
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190
>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>>
>> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
>> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel
>> sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>>
>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel
>> sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
>>
>> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
>> confirmed.
>>
>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
>> 2001)
>>
>> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848
>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>>
>> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup
>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
>> file
>> /usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.wav
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
>> sofia/internal/102 at 192.168.1.6:5060 skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
>> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel
>> sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
>> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: NORMAL_CLEARING
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> CS_REPORTING
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>>
>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
>> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
>> NORMAL_CLEARING
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
>> CS_DESTROY
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
>> (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
>>
>> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
>> (sofia/internal/102 at 192.168.1.6:5060) Ended
>>
>> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close
>> Channel sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374
>> sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
>> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>>
>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>>
>> freeswitch at internal> /bye
>>
>> [root at server bin]#
>>
>> Regards,
>>
>> Phil Quesinberry
>>
>> Q Systems Engineering, Inc.
>>
>> Electronic Controls and Embedded Systems Development
>>
>> (410) 969-8002
>>
>> http://www.qsystemsengineering.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Apr  2 20:38:07 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 11:38:07 -0500
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <CAKbxfG9P_B9BYzh9QCx+GUuv5uWw1pC_WeYcXW29Gr9Z0+S+xA@mail.gmail.com>
References: <000301cd108a$4d3f55a0$e7be00e0$@com>
	<CANBhyPBtJC5uBxZnC=YtWCj5AFx-g9imD=O3WzKiUQZ5sDQEBw@mail.gmail.com>
	<-82712645424086336@unknownmsgid>
	<CAKbxfG9P_B9BYzh9QCx+GUuv5uWw1pC_WeYcXW29Gr9Z0+S+xA@mail.gmail.com>
Message-ID: <CAKbxfG88_YAev8oJCEx5Hew25OiSrrAjLV7mvfjv5RPtkzR47A@mail.gmail.com>

plus the log is not detailed enough

sofia global siptrace on
sofia loglevel all 9
console loglevel debug (/log debug)

you would need enough log to cover the time when the phone is working
until when it stops working.

On Mon, Apr 2, 2012 at 11:35 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> and use pastebin.freeswitch.org or attachments for logs imagine
> reading this from your mobile phone.
>
> On Mon, Apr 2, 2012 at 1:26 AM, Brian West <brian at freeswitch.org> wrote:
>> And use Jira to report bugs please.
>>
>> /b
>>
>> Sent from my iPhone
>>
>> On Apr 2, 2012, at 1:20 AM, Anton Kvashenkin <anton.jugatsu at gmail.com>
>> wrote:
>>
>> I think that the best way to check endpoints registrations is sofia status
>> profile internal reg. I have sometimes accuracy problems with show
>> registrations.
>>
>> 2 ?????? 2012 ?. 8:37 ???????????? Phil Quesinberry
>> <philq at qsystemsengineering.com> ???????:
>>>
>>> Hi,
>>>
>>> I've noticed what appears to be a bug which I wanted to bring to your
>>> attention. ?Sometimes when FS is started, extensions can't be called even
>>> after they've registered. ?This doesn't happen all the time... and restarting
>>> FS always solves the problem. ?Just to be clear - this problem only
>>> sometimes occurs when FS has been started/restarted, like after an update.
>>> Once FS has "started successfully" and sees presence for registered
>>> extensions, it continues to work fine.
>>>
>>> It's all or nothing - no extensions can be dialed when this issue is
>>> present, whether they're local or NATted. ?If one extension can be reached,
>>> you can reach them all (if they're registered, of course).
>>>
>>> See debug info below on an attempt to dial a local (no NAT) extension -
>>> this was one of the recent beta git pulls and the problem has been present
>>> through several updates. ?Let me know if you need any additional info.
>>>
>>>
>>> freeswitch at internal> show registrations
>>>
>>>
>>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>>>
>>>
>>> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>>>
>>>
>>> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.182:5060;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>>>
>>>
>>> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>>>
>>>
>>> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.222.182:1066;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>>>
>>>
>>> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.222.182:1068;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>>>
>>>
>>> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.222.182:1067;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>>>
>>>
>>> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.100.94:54629;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>>>
>>>
>>> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.100.94:6480;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>>>
>>>
>>> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.222.182:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>>>
>>> 9 total.
>>>
>>> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
>>> acl "domains". Falling back to Digest auth.
>>>
>>> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
>>> (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
>>> 192.168.1.4
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
>>> acl "domains". Falling back to Digest auth.
>>>
>>> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
>>> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel
>>> sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>>>
>>> v=0
>>>
>>> o=MxSIP 0 1 IN IP4 192.168.1.4
>>>
>>> s=SIP Call
>>>
>>> c=IN IP4 192.168.1.4
>>>
>>> t=0 0
>>>
>>> m=audio 16384 RTP/AVP 0 110 8 9
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> a=rtpmap:110 PCMU/16000
>>>
>>> a=rtpmap:8 PCMA/8000
>>>
>>> a=rtpmap:9 G722/8000
>>>
>>> a=silenceSupp:off - - - -
>>>
>>> a=ptime:20
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746
>>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85
>>> sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125
>>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
>>> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886
>>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
>>> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148
>>> sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
>>> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>>>
>>> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
>>> <102>->2001 in context default
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->call_direction] continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
>>> set(call_direction=local)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->tod_example] continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
>>> [tod_example] break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->holiday_example] continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
>>> [holiday_example] break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->group-intercept] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [group-intercept] destination_number(2001) =~ /^\*8$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
>>> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
>>> continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
>>> ${call_debug}(false) =~ /^true$/ break=never
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
>>> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
>>> break=never
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-last_dial/global/${uuid})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->snom-demo-2] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
>>> destination_number(2001) =~ /^\*9001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->snom-demo-1] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
>>> destination_number(2001) =~ /^\*9000$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->call_privacy] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
>>> destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->call_return] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
>>> destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->del-group]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
>>> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->add-group]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
>>> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->call-group-simo] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [call-group-simo] destination_number(2001) =~ /^\*\*82(\d{2})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->call-group-order] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->intercept-ext] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [intercept-ext]
>>> destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Local_Extension_Skinny] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->send_to_voicemail] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->enum.9.10d] continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
>>> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->enum.9.11d] continue=true
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
>>> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->SipBroker.SIPDial] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->SipBroker.Toll_Free] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [SipBroker.Toll_Free] destination_number(2001) =~
>>> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->TollFreeGateway] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [TollFreeGateway] destination_number(2001) =~
>>> /^91?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->FlowRoute.91d10or11] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->HL7519.9.10d] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
>>> caller_id_number(102) =~ /519/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->HL7519.9.11d] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
>>> caller_id_number(102) =~ /519/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->HL7612.9.10d] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
>>> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
>>> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->HL7612.9.11d] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
>>> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
>>> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
>>> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d10]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
>>> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d11]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
>>> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>>> Clock Time] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>>> Time] destination_number(2001) =~ /^9170$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>>> Clock Date] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>>> Date] destination_number(2001) =~ /^9171$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
>>> Clock Date and Time] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
>>> Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Recordings] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
>>> destination_number(2001) =~ /^\*(732)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Company_Directory] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Hersch_Lauren_Main] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Holiday_Announcement] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
>>> admin phones.park] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
>>> phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
>>> admin phones] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
>>> phones] destination_number(2001) =~ /^1001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->group_dial_sales] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->group_dial_support] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->group_dial_billing] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
>>> destination_number(2001) =~ /^operator$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
>>> destination_number(2001) =~ /^vmain$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
>>> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
>>> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
>>> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->nb_conferences] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [nb_conferences] destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->wb_conferences] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [wb_conferences] destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->uwb_conferences] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [uwb_conferences] destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->cdquality_conferences] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->freeswitch_public_conf_via_sip] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [freeswitch_public_conf_via_sip] destination_number(2001) =~
>>> /^\*9(888|8888|1616|3232)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
>>> destination_number(2001) =~ /^\*5000$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->dynamic_conference] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->rtp_multicast_page] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->parking_slots] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [parking_slots]
>>> destination_number(2001) =~ /^(\*59[0-9][0-9])$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->valet_park_in] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_in]
>>> destination_number(2001) =~ /^\*(6000)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->valet_park_out] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [valet_park_out] destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
>>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
>>> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
>>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
>>> destination_number(2001) =~ /^parking$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
>>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
>>> destination_number(2001) =~ /callpark/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
>>> source(mod_sofia) =~ /mod_sofia/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
>>> destination_number(2001) =~ /pickup/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
>>> destination_number(2001) =~ /^wait$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->fax_receive] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
>>> destination_number(2001) =~ /^\*9178$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->fax_transmit] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
>>> destination_number(2001) =~ /^\*9179$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->ringback_180] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
>>> destination_number(2001) =~ /^\*9180$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->ringback_183_uk_ring] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->ringback_183_music_ring] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->ringback_post_answer_uk_ring] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->ringback_post_answer_music] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
>>> break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
>>> destination_number(2001) =~ /^\*9191$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->show_info]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
>>> destination_number(2001) =~ /^\*9192$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->video_record] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
>>> destination_number(2001) =~ /^\*9193$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->video_playback] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->delay_echo] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
>>> destination_number(2001) =~ /^\*9195$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
>>> destination_number(2001) =~ /^\*9196$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->milliwatt]
>>> continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
>>> destination_number(2001) =~ /^\*9197$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->tone_stream] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
>>> destination_number(2001) =~ /^\*9198$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->zrtp_enrollement] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
>>> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->hold_music] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
>>> destination_number(2001) =~ /^\*9664$/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
>>> [default->Local_Extension] continue=false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS)
>>> [Local_Extension] destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(dialed_extension=2001)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> export(dialed_extension=2001)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db
>>> ${domain_name} 2001 ${limit_max} ${limit_destination})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
>>> execute_extension::dx XML features)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
>>> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
>>> execute_extension::cf XML features)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
>>> execute_extension::att_xfer XML features)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
>>> 4000, 440.0, 480.0))
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(hangup_after_bridge=true)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(continue_on_fail=true)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
>>> var callgroup)})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
>>> bridge(user/${user_data(${destination_number}@${domain_name} attr
>>> id)}@${domain_name})
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>>>
>>> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
>>> ${domain_name} ${dialed_extension})
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
>>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> CS_EXECUTE
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
>>> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
>>> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241
>>> sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
>>> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-last_dial/102/2001)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
>>> 2012 23:50:41 -0400)
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012
>>> 23:50:41 -0400]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>>>
>>> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
>>> (export_vars) [dialed_extension]=[2001]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>>>
>>> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
>>> 192.168.1.6_2001 max:5, interval:0
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001
>>> is now 1/5
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
>>> execute_extension::dx XML features)
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
>>> execute_extension::dx XML features
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
>>> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
>>> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
>>> execute_extension::cf XML features)
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
>>> execute_extension::cf XML features
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
>>> execute_extension::att_xfer XML features)
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
>>> execute_extension::att_xfer XML features
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
>>> 440.0, 480.0))
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0,
>>> 480.0)]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
>>> 4000, 440.0, 480.0))
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000,
>>> 440.0, 480.0)]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-call_return/2001/102)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
>>> sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>>> hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
>>> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
>>> [dialed_extension]=[2001] to event
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
>>> global variables
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
>>> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
>>> [dialed_extension]=[2001] to event
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
>>> global variables
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
>>> [sip_invite_domain]=[192.168.1.6]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
>>> [presence_id]=[2001 at 192.168.1.6]
>>>
>>> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
>>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
>>> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>>>
>>> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
>>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
>>> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>>>
>>> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
>>> Cause: USER_NOT_REGISTERED
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
>>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
>>> Disable 2833 dtmf and switch to INFO
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
>>> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 192.168.1.4
>>> port 16384 codec: 0 ms: 20
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
>>> 160 bytes per 20ms
>>>
>>> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
>>> sofia/internal/102 at 192.168.1.6:5060!
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932
>>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
>>> sofia/internal/102 at 192.168.1.6:5060:
>>>
>>> v=0
>>>
>>> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>>>
>>> s=FreeSWITCH
>>>
>>> c=IN IP4 192.168.1.6
>>>
>>> t=0 0
>>>
>>> m=audio 23164 RTP/AVP 0
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> a=silenceSupp:off - - - -
>>>
>>> a=ptime:20
>>>
>>> a=sendrecv
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190
>>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>>>
>>> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
>>> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel
>>> sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>>>
>>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel
>>> sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
>>>
>>> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
>>> confirmed.
>>>
>>> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
>>> 2001)
>>>
>>> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>>
>>> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848
>>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>>>
>>> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup
>>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
>>> file
>>> /usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.wav
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
>>> sofia/internal/102 at 192.168.1.6:5060 skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
>>> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel
>>> sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
>>> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: NORMAL_CLEARING
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
>>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> CS_REPORTING
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>>>
>>> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
>>> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
>>> NORMAL_CLEARING
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
>>> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
>>> CS_DESTROY
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
>>> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
>>> (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
>>>
>>> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
>>> (sofia/internal/102 at 192.168.1.6:5060) Ended
>>>
>>> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close
>>> Channel sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
>>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
>>> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374
>>> sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
>>> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>>>
>>> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>>>
>>> freeswitch at internal> /bye
>>>
>>> [root at server bin]#
>>>
>>> Regards,
>>>
>>> Phil Quesinberry
>>>
>>> Q Systems Engineering, Inc.
>>>
>>> Electronic Controls and Embedded Systems Development
>>>
>>> (410) 969-8002
>>>
>>> http://www.qsystemsengineering.com
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Apr  2 22:08:24 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 11:08:24 -0700
Subject: [Freeswitch-users] Internal profile not started at FS startup
 if extension uses TCP
In-Reply-To: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
Message-ID: <CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>

What is the 2am restart? Is that a cron job or...?

On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com> wrote:

> I added an iPad softphone that requires TCP to converse battery.
> Everything works except: FS is restarted 2am each day but does not activate
> the INTERNAL profile, this causes all internal extensions to stay offline.
> If  the TCP extension is off all works. Odd: I cannot reproduce this at any
> other time manually, it only occurs at the 2am restart! So debugging has
> been a bugger. Anytime after 2am I can "sofia profile internal start" and
> all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did
> not show up in the log at all and NO MENTION of INTERNAL in the log. It's
> as if the file is missing, there is nothing in the FS log to provide a
> hint. I searched the FS wiki and web and found a TCP bug mentioned march
> 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway)
> and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and
> both had the problem. Anyone have an idea what to try next?
> Mario G
>
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From msc at freeswitch.org  Mon Apr  2 22:26:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 11:26:45 -0700
Subject: [Freeswitch-users] ClueCon 2012 Registration Now Open!
Message-ID: <CAKzWOxUHapMQJCOj6=-T20GDU=FeYEzF26KoW=Mg-PJTtN=+Vw@mail.gmail.com>

*ClueCon 2012 - Register Today!*

The ClueCon team is pleased to announce that registration is now open for
ClueCon 2012. We invite all to visit our new
website<http://www.cluecon.com/?cc12reg01>and get registered today.
2012 marks the 8th annual ClueCon event. This
year we return to the beautiful Wyndham Hotel in downtown Chicago. ClueCon
2012 will be held August 7-9.

ClueCon is best known as the open source telephony conference that is by
developers, for developers. As in years past we will have highly technical
presentations on a variety of subjects. You'll be able to rub elbows with
the authors of powerful open source telephony projects like Kamailio,
OpenSIPS, and, of course, FreeSWITCH. VoIP security will again be an
important subject and we are pleased to have Phil Zimmermann returning to
discuss how VoIP security affects us all and how ZRTP can help.

Stay tuned for announcements about specific talks, speakers, new sponsors,
and the ClueCon 2012 schedule of events.

*Wyndham Discount
*

Those staying at the Wyndham will enjoy a $300 discount off the regular
admission price of $999. Select the appropriate PayPal button from the
registration <http://www.cluecon.com/register/?cc12reg01> page. For more
information about the Wyndham, visit the ClueCon
hotel<http://www.cluecon.com/hotels/?cc12reg01>page.

Make your plans and register right away. We look forward to seeing everyone
this August!

The ClueCon Team
http://www.ClueCon.com <http://www.cluecon.com/>
877.742.CLUE

 <http://www.cluecon.com/?cc12reg01>
CC120402-1
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From mario_fs at mgtech.com  Mon Apr  2 22:30:58 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 11:30:58 -0700
Subject: [Freeswitch-users] Internal profile not started at FS startup
	if extension uses TCP
In-Reply-To: <CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
	<CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>
Message-ID: <EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>

It is a plist on OSX. Has been fine since 2010. I duplicated the issue by setting the in the plist, here is a replacement since I turned on all kinds of logging and finally found a message. Here is an update:

A TCP extension causes already in use MSG and  Internal profile fails

I added an iPad softphone (tested SoftPhone and Bria) that requires TCP and all works fine except: When FS is restarted 2am each day the INTERNAL profile is not activated causing all internal extensions to stay offline. If  the TCP extension is off or set UDP all works.  Anytime after the restart I can "sofia profile internal start" to fix it. I am on Saturdays (03-30) GIT (to test the new 1.2). The only thing I can find is the "Address already in use" below which is shortly followed by the error creating internal. The .7 address is FS and there is nothing else using it. Any ideas why it says in use ONLY when an extension is set for TCP?
Mario G

tport_bind_server(0x1003a9850): calling tport_listen for tcp
nta: master transport created
tport_alloc_primary(0x1003a9850): new primary tport 0x1003993e0
tport_listen(0x1003a9850): bind(pf=2 tcp/[100.234.1.7]:5060): Address already in use   <---------------
nta: bind(100.234.1.7:5060;transport=*): Address already in use
nua: initializing SIP stack failed
nua: nua_stack_deinit: entering
tport_bind_server(0x1006dd850) to */100.234.1.7:5078/sip
sres_sofia_update(0x1003a7e10, -1, -1)
sres_sofia_update(0x0, -1, -1)
tport_destroy(0x1003a9850)
tport_bind_server(0x1006dd850): calling tport_listen for udp
tport_alloc_primary(0x1006dd850): new primary tport 0x1003a9ba0
su_kqueue_port_deinit(0x10039e350) called
tport_listen(0x1003a9ba0): listening at udp/100.234.1.7:5078/sip
tport_bind_server(0x1006dd850): calling tport_listen for tcp
tport_alloc_primary(0x1006dd850): new primary tport 0x10039e440
tport_listen(0x10039e440): listening at tcp/100.234.1.7:5078/sip
nta: bound to (100.234.1.7:5078;transport=*)
nta: agent_init_via: SIP/2.0/udp 100.234.1.7:5078 (sip)
nta: agent_init_via: SIP/2.0/tcp 100.234.1.7:5078 (sip)
nta: Via fields initialized
nta: Contact header created
2012-04-02 11:00:35.147184 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal   <---------------------
nua_register: Adding contact URL '100.234.1.7' to list.
tport_wakeup_pri(0x1003a9ba0): events IN
tport_recv_event(0x1003a9ba0)
tport_recv_iovec(0x1003a9ba0) msg 0x1003a74e0 from (udp/100.234.1.7:5078) has 20 bytes, veclen = 1




On Apr 2, 2012, at 11:08 AM, Michael Collins wrote:

> What is the 2am restart? Is that a cron job or...?
> 
> On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com> wrote:
> I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next?
> Mario G
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From peter.olsson at visionutveckling.se  Mon Apr  2 22:47:08 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 2 Apr 2012 18:47:08 +0000
Subject: [Freeswitch-users] Internal profile not started at FS
	startup	if extension uses TCP
In-Reply-To: <EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
	<CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>,
	<EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>
Message-ID: <0AB00DEE-88D8-44E8-9F34-00A1218565BC@visionutveckling.se>

It means port 5060 is busy (probably by the old FS instance). How do you restart FS?

/Peter

2 apr 2012 kl. 20:34 skrev "Mario G" <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>>:

It is a plist on OSX. Has been fine since 2010. I duplicated the issue by setting the in the plist, here is a replacement since I turned on all kinds of logging and finally found a message. Here is an update:

A TCP extension causes already in use MSG and  Internal profile fails

I added an iPad softphone (tested SoftPhone and Bria) that requires TCP and all works fine except: When FS is restarted 2am each day the INTERNAL profile is not activated causing all internal extensions to stay offline. If  the TCP extension is off or set UDP all works.  Anytime after the restart I can "sofia profile internal start" to fix it. I am on Saturdays (03-30) GIT (to test the new 1.2). The only thing I can find is the "Address already in use" below which is shortly followed by the error creating internal. The .7 address is FS and there is nothing else using it. Any ideas why it says in use ONLY when an extension is set for TCP?
Mario G

tport_bind_server(0x1003a9850): calling tport_listen for tcp
nta: master transport created
tport_alloc_primary(0x1003a9850): new primary tport 0x1003993e0
tport_listen(0x1003a9850): bind(pf=2 tcp/[100.234.1.7]:5060): Address already in use   <---------------
nta: bind(100.234.1.7:5060;transport=*): Address already in use
nua: initializing SIP stack failed
nua: nua_stack_deinit: entering
tport_bind_server(0x1006dd850) to */100.234.1.7:5078/sip
sres_sofia_update(0x1003a7e10, -1, -1)
sres_sofia_update(0x0, -1, -1)
tport_destroy(0x1003a9850)
tport_bind_server(0x1006dd850): calling tport_listen for udp
tport_alloc_primary(0x1006dd850): new primary tport 0x1003a9ba0
su_kqueue_port_deinit(0x10039e350) called
tport_listen(0x1003a9ba0): listening at udp/100.234.1.7:5078/sip
tport_bind_server(0x1006dd850): calling tport_listen for tcp
tport_alloc_primary(0x1006dd850): new primary tport 0x10039e440
tport_listen(0x10039e440): listening at tcp/100.234.1.7:5078/sip
nta: bound to (100.234.1.7:5078;transport=*)
nta: agent_init_via: SIP/2.0/udp 100.234.1.7:5078 (sip)
nta: agent_init_via: SIP/2.0/tcp 100.234.1.7:5078 (sip)
nta: Via fields initialized
nta: Contact header created
2012-04-02 11:00:35.147184 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal   <---------------------
nua_register: Adding contact URL '100.234.1.7' to list.
tport_wakeup_pri(0x1003a9ba0): events IN
tport_recv_event(0x1003a9ba0)
tport_recv_iovec(0x1003a9ba0) msg 0x1003a74e0 from (udp/100.234.1.7:5078) has 20 bytes, veclen = 1




On Apr 2, 2012, at 11:08 AM, Michael Collins wrote:

What is the 2am restart? Is that a cron job or...?

On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>> wrote:
I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next?
Mario G

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f79ee9c32766092896665!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4f79ee9c32766092896665!


From Mailings at kh-dev.de  Mon Apr  2 22:54:42 2012
From: Mailings at kh-dev.de (Klaus Hochlehnert)
Date: Mon, 2 Apr 2012 18:54:42 +0000
Subject: [Freeswitch-users] Playing additional file when transferring call
Message-ID: <8220BA48EF3C9A438124F20C9C76571520EE6782@srv01.khdev.corp>

Hi,

I have a question if this is possible and how:
When transferring a call (but only when it was picked up and from an external source) I'd like to play a complete sound file till the end before going to "transfer_ringback".

Does anyone have an idea how to do that?

Thanks
Klaus
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From mario_fs at mgtech.com  Mon Apr  2 22:59:34 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 11:59:34 -0700
Subject: [Freeswitch-users] Internal profile not started at FS
	startup	if extension uses TCP
In-Reply-To: <0AB00DEE-88D8-44E8-9F34-00A1218565BC@visionutveckling.se>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
	<CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>,
	<EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>
	<0AB00DEE-88D8-44E8-9F34-00A1218565BC@visionutveckling.se>
Message-ID: <8103D182-9384-4861-9A1E-27D03E53EF68@mgtech.com>

There is a plist that starts FS when it is down, then another plist shuts it down once a day, the startup plist causes it to restart. It's exactly the ones I put in the FS OSX install wiki (speaking of I will update this week). It's odd that it ONLY happens when an extension is set to TCP. Also, I am assuming this is the problem... I will have to retest without the TCP extensions to see it it is.

On Apr 2, 2012, at 11:47 AM, Peter Olsson wrote:

> It means port 5060 is busy (probably by the old FS instance). How do you restart FS?
> 
> /Peter
> 
> 2 apr 2012 kl. 20:34 skrev "Mario G" <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>>:
> 
> It is a plist on OSX. Has been fine since 2010. I duplicated the issue by setting the in the plist, here is a replacement since I turned on all kinds of logging and finally found a message. Here is an update:
> 
> A TCP extension causes already in use MSG and  Internal profile fails
> 
> I added an iPad softphone (tested SoftPhone and Bria) that requires TCP and all works fine except: When FS is restarted 2am each day the INTERNAL profile is not activated causing all internal extensions to stay offline. If  the TCP extension is off or set UDP all works.  Anytime after the restart I can "sofia profile internal start" to fix it. I am on Saturdays (03-30) GIT (to test the new 1.2). The only thing I can find is the "Address already in use" below which is shortly followed by the error creating internal. The .7 address is FS and there is nothing else using it. Any ideas why it says in use ONLY when an extension is set for TCP?
> Mario G
> 
> tport_bind_server(0x1003a9850): calling tport_listen for tcp
> nta: master transport created
> tport_alloc_primary(0x1003a9850): new primary tport 0x1003993e0
> tport_listen(0x1003a9850): bind(pf=2 tcp/[100.234.1.7]:5060): Address already in use   <---------------
> nta: bind(100.234.1.7:5060;transport=*): Address already in use
> nua: initializing SIP stack failed
> nua: nua_stack_deinit: entering
> tport_bind_server(0x1006dd850) to */100.234.1.7:5078/sip
> sres_sofia_update(0x1003a7e10, -1, -1)
> sres_sofia_update(0x0, -1, -1)
> tport_destroy(0x1003a9850)
> tport_bind_server(0x1006dd850): calling tport_listen for udp
> tport_alloc_primary(0x1006dd850): new primary tport 0x1003a9ba0
> su_kqueue_port_deinit(0x10039e350) called
> tport_listen(0x1003a9ba0): listening at udp/100.234.1.7:5078/sip
> tport_bind_server(0x1006dd850): calling tport_listen for tcp
> tport_alloc_primary(0x1006dd850): new primary tport 0x10039e440
> tport_listen(0x10039e440): listening at tcp/100.234.1.7:5078/sip
> nta: bound to (100.234.1.7:5078;transport=*)
> nta: agent_init_via: SIP/2.0/udp 100.234.1.7:5078 (sip)
> nta: agent_init_via: SIP/2.0/tcp 100.234.1.7:5078 (sip)
> nta: Via fields initialized
> nta: Contact header created
> 2012-04-02 11:00:35.147184 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal   <---------------------
> nua_register: Adding contact URL '100.234.1.7' to list.
> tport_wakeup_pri(0x1003a9ba0): events IN
> tport_recv_event(0x1003a9ba0)
> tport_recv_iovec(0x1003a9ba0) msg 0x1003a74e0 from (udp/100.234.1.7:5078) has 20 bytes, veclen = 1
> 
> 
> 
> 
> On Apr 2, 2012, at 11:08 AM, Michael Collins wrote:
> 
> What is the 2am restart? Is that a cron job or...?
> 
> On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>> wrote:
> I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next?
> Mario G
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4f79ee9c32766092896665!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> !DSPAM:4f79ee9c32766092896665!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Mon Apr  2 23:41:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 12:41:16 -0700
Subject: [Freeswitch-users] ZRTP Code Now In Tree
Message-ID: <CAKzWOxVrGoJhfroAHoLQMa09SKFQQiwSmdywRH7CghNHYU+wmA@mail.gmail.com>

Hello all!

This isn't an announcement so much as an FYI. Phil Zimmermann has been
working diligently with our very own power user Travis Cross (IRC: tc) on
getting ZRTP into a truly useful state. The first big step on that journey
has been taken: libzrtp code has been added to the git repository. Travis
Cross reports that getting to this point was a serious amount of work, and
for that we thank him and Phil both. We are all going to benefit from ZRTP
when it is ready for use. The code is available for your perusal, however
it is not quite ready for use at this time.

We invite everyone to join us on the FreeSWITCH conference call this
Wednesday (agenda here<http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_04>).
Travis will be discussing what has been done thus far and what remains to
be done. This will dovetail nicely with AEP who will be discussing the
Freedom Fone project on Wednesday also.

-Michael
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From mario_fs at mgtech.com  Mon Apr  2 23:42:35 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 12:42:35 -0700
Subject: [Freeswitch-users] Internal profile not started at FS
	startup	if extension uses TCP
In-Reply-To: <8103D182-9384-4861-9A1E-27D03E53EF68@mgtech.com>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
	<CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>,
	<EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>
	<0AB00DEE-88D8-44E8-9F34-00A1218565BC@visionutveckling.se>
	<8103D182-9384-4861-9A1E-27D03E53EF68@mgtech.com>
Message-ID: <8454EDB4-1A55-4D10-9B04-1E886E976B23@mgtech.com>

Looks like I know what it is but have no idea how to fix it. Multiple tests are reproducible. I did a NETSTAT -p TCP to watch the .7 address (oddly port 5060 show up as "sip" in the netstat command on OSX). When FS comes down this port stays open for about 30 seconds which is longer than the FS restart takes so sofia says .7:5060 is in use. I also tied setting keep alive to 0. I now finally figured out the problem but.... Does anyone have an idea how to force the ip/port closed sooner? 

On Apr 2, 2012, at 11:59 AM, Mario G wrote:

> There is a plist that starts FS when it is down, then another plist shuts it down once a day, the startup plist causes it to restart. It's exactly the ones I put in the FS OSX install wiki (speaking of I will update this week). It's odd that it ONLY happens when an extension is set to TCP. Also, I am assuming this is the problem... I will have to retest without the TCP extensions to see it it is.
> 
> On Apr 2, 2012, at 11:47 AM, Peter Olsson wrote:
> 
>> It means port 5060 is busy (probably by the old FS instance). How do you restart FS?
>> 
>> /Peter
>> 
>> 2 apr 2012 kl. 20:34 skrev "Mario G" <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>>:
>> 
>> It is a plist on OSX. Has been fine since 2010. I duplicated the issue by setting the in the plist, here is a replacement since I turned on all kinds of logging and finally found a message. Here is an update:
>> 
>> A TCP extension causes already in use MSG and  Internal profile fails
>> 
>> I added an iPad softphone (tested SoftPhone and Bria) that requires TCP and all works fine except: When FS is restarted 2am each day the INTERNAL profile is not activated causing all internal extensions to stay offline. If  the TCP extension is off or set UDP all works.  Anytime after the restart I can "sofia profile internal start" to fix it. I am on Saturdays (03-30) GIT (to test the new 1.2). The only thing I can find is the "Address already in use" below which is shortly followed by the error creating internal. The .7 address is FS and there is nothing else using it. Any ideas why it says in use ONLY when an extension is set for TCP?
>> Mario G
>> 
>> tport_bind_server(0x1003a9850): calling tport_listen for tcp
>> nta: master transport created
>> tport_alloc_primary(0x1003a9850): new primary tport 0x1003993e0
>> tport_listen(0x1003a9850): bind(pf=2 tcp/[100.234.1.7]:5060): Address already in use   <---------------
>> nta: bind(100.234.1.7:5060;transport=*): Address already in use
>> nua: initializing SIP stack failed
>> nua: nua_stack_deinit: entering
>> tport_bind_server(0x1006dd850) to */100.234.1.7:5078/sip
>> sres_sofia_update(0x1003a7e10, -1, -1)
>> sres_sofia_update(0x0, -1, -1)
>> tport_destroy(0x1003a9850)
>> tport_bind_server(0x1006dd850): calling tport_listen for udp
>> tport_alloc_primary(0x1006dd850): new primary tport 0x1003a9ba0
>> su_kqueue_port_deinit(0x10039e350) called
>> tport_listen(0x1003a9ba0): listening at udp/100.234.1.7:5078/sip
>> tport_bind_server(0x1006dd850): calling tport_listen for tcp
>> tport_alloc_primary(0x1006dd850): new primary tport 0x10039e440
>> tport_listen(0x10039e440): listening at tcp/100.234.1.7:5078/sip
>> nta: bound to (100.234.1.7:5078;transport=*)
>> nta: agent_init_via: SIP/2.0/udp 100.234.1.7:5078 (sip)
>> nta: agent_init_via: SIP/2.0/tcp 100.234.1.7:5078 (sip)
>> nta: Via fields initialized
>> nta: Contact header created
>> 2012-04-02 11:00:35.147184 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal   <---------------------
>> nua_register: Adding contact URL '100.234.1.7' to list.
>> tport_wakeup_pri(0x1003a9ba0): events IN
>> tport_recv_event(0x1003a9ba0)
>> tport_recv_iovec(0x1003a9ba0) msg 0x1003a74e0 from (udp/100.234.1.7:5078) has 20 bytes, veclen = 1
>> 
>> 
>> 
>> 
>> On Apr 2, 2012, at 11:08 AM, Michael Collins wrote:
>> 
>> What is the 2am restart? Is that a cron job or...?
>> 
>> On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>> wrote:
>> I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next?
>> Mario G
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> !DSPAM:4f79ee9c32766092896665!
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> !DSPAM:4f79ee9c32766092896665!
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From philq at qsystemsengineering.com  Mon Apr  2 23:44:22 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Mon, 02 Apr 2012 15:44:22 -0400
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
Message-ID: <020701cd1109$037d8cf0$0a78a6d0$@com>

The phone never stops working.  The problem is either present from the time
FS is started or it isn't.  I'll have to bring it down and try to get it to
fail again after hours so I can provide more detailed info.  To me, it
appears from the log I captured that presence_id is not always being
initialized properly when FS is started but I haven't looked at the source
code to see what's really happening.  I do have the two FS books on the way
though, which should give me a deeper understanding of how things work.  The
wiki only goes so far.

Also, when I do a sofia status profile internal reg, I see a bunch of long
expired registration entries.  Not sure if that could somehow be related to
this problem.


freeswitch at internal> sofia status profile internal reg

Registrations:
============================================================================
=====================
Call-ID:        49e008f79bd41460
User:           102 at 192.168.1.6
Contact:        "Phil" <sip:102 at 192.168.1.4:5060;transport=udp>
Agent:          Aastra 9480i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:39:06)
EXPSECS(271)
Host:           Server
IP:             192.168.1.4
Port:           5060
Auth-User:      102
Auth-Realm:     192.168.1.6
MWI-Account:    102 at 192.168.1.6

Call-ID:        5f92c72e93a50f67
User:           226 at 192.168.1.6
Contact:        "Sonny" <sip:226 at 74.93.xx.xx:5060;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:49:47)
EXPSECS(-2681088)
Host:           Server
IP:             74.93.xx.xx
Port:           1066
Auth-User:      226
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    226 at 192.168.1.6

Call-ID:        09856e1e62e97740
User:           227 at 192.168.1.6
Contact:        "Laura" <sip:227 at 74.93.xx.xx:1064;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:47:39)
EXPSECS(-2681216)
Host:           Server
IP:             74.93.xx.xx
Port:           5060
Auth-User:      227
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    227 at 192.168.1.6

Call-ID:        69748b87cb4b8d1d
User:           257 at 192.168.1.6
Contact:        "Tina" <sip:257 at 74.93.xx.xx:1060;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:49:11)
EXPSECS(-2681124)
Host:           Server
IP:             74.93.xx.xx
Port:           1064
Auth-User:      257
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    257 at 192.168.1.6

Call-ID:        45249b30-85ddd13a at 10.0.0.159
User:           519 at 192.168.1.6
Contact:        Fax Machine <sip:519 at 74.93.xx.xx:1062>
Agent:          Linksys/SPA2102-5.2.12
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:47)
EXPSECS(132)
Host:           Server
IP:             74.93.xx.xx
Port:           1062
Auth-User:      519
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    519 at 192.168.1.6

Call-ID:        effe7aec41953215
User:           102 at 192.168.2.105
Contact:        "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
Agent:          Aastra 9480i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-03-09 00:22:38)
EXPSECS(-2124717)
Host:           localhost.localdomain
IP:             192.168.2.104
Port:           5060
Auth-User:      102
Auth-Realm:     192.168.2.105
MWI-Account:    102 at 192.168.2.105

Call-ID:        e9351a6c40675c86
User:           102 at 192.168.2.6
Contact:        "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
Agent:          Aastra 9480i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-03-15 00:46:48)
EXPSECS(-1608467)
Host:           pbx
IP:             192.168.2.104
Port:           5060
Auth-User:      102
Auth-Realm:     192.168.2.6
MWI-Account:    102 at 192.168.2.6

Call-ID:        45249b30-85ddd13a at 10.0.0.159
User:           519 at 192.168.2.6
Contact:        Fax Machine <sip:519 at 74.93.xx.xx:1062>
Agent:          Linksys/SPA2102-5.2.12
Status:         Registered(UDP)(unknown) EXP(2012-03-15 00:46:23)
EXPSECS(-1608492)
Host:           pbx
IP:             74.93.xx.xx
Port:           1062
Auth-User:      519
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    519 at 192.168.2.6

Call-ID:        864e7aef4c59c423
User:           225 at 192.168.1.6
Contact:        "Vicky" <sip:225 at 74.93.xx.xx:1067;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:25)
EXPSECS(110)
Host:           server
IP:             74.93.xx.xx
Port:           1067
Auth-User:      225
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    225 at 192.168.1.6

Call-ID:        29b0d19d8063e55c
User:           226 at 192.168.1.6
Contact:        "Sonny" <sip:226 at 74.93.xx.xx:1068;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:25)
EXPSECS(110)
Host:           server
IP:             74.93.xx.xx
Port:           1068
Auth-User:      226
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    226 at 192.168.1.6

Call-ID:        6bc00b660278e069
User:           227 at 192.168.1.6
Contact:        "Laura" <sip:227 at 74.93.xx.xx:1066;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:35:51)
EXPSECS(76)
Host:           server
IP:             74.93.xx.xx
Port:           1066
Auth-User:      227
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    227 at 192.168.1.6

Call-ID:        e721049935e82279
User:           257 at 192.168.1.6
Contact:        "Tina" <sip:257 at 74.93.xx.xx:5060;transport=udp>
Agent:          Aastra 9143i/3.2.2.1136
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:32)
EXPSECS(117)
Host:           server
IP:             74.93.xx.xx
Port:           5060
Auth-User:      257
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    257 at 192.168.1.6

Call-ID:        54667b49ca74cabe
User:           3900 at 192.168.1.6
Contact:        "Brad"
<sip:3900 at 173.66.xx.xx:54629;transport=udp;fs_nat=yes>
Agent:          Aastra 9143i/3.2.2.2044
Status:         Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:22)
EXPSECS(107)
Host:           server.home
IP:             173.66.xx.xx
Port:           54629
Auth-User:      3900
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    3900 at 192.168.1.6

Call-ID:        6dc54f7994ffb61a
User:           3901 at 192.168.1.6
Contact:        "KCC Main"
<sip:3901 at 173.66.xx.xx:6480;transport=udp;fs_nat=yes>
Agent:          Aastra 9143i/3.2.2.2044
Status:         Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:38)
EXPSECS(123)
Host:           server.home
IP:             173.66.xx.xx
Port:           6480
Auth-User:      3901
Auth-Realm:     qsystemseng.ourdomain.com
MWI-Account:    3901 at 192.168.1.6

Call-ID:        bae1af9975ebfa50
User:           2001 at 192.168.1.6
Contact:        "Phil" <sip:2001 at 192.168.1.4:5060;transport=udp>
Agent:          Aastra 9480i/3.2.2.2044
Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:37)
EXPSECS(122)
Host:           server.home
IP:             192.168.1.4
Port:           5060
Auth-User:      2001
Auth-Realm:     192.168.1.6
MWI-Account:    2001 at 192.168.1.6

Total items returned: 15
============================================================================
=====================



- Phil

----------
Anthony Minessale wrote:
plus the log is not detailed enough

sofia global siptrace on
sofia loglevel all 9
console loglevel debug (/log debug)

you would need enough log to cover the time when the phone is working
until when it stops working.


_____________________________________________
From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] 
Sent: Monday, April 02, 2012 12:37 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Possible registration/presence bug - Sometimes get "Originate
Failed. Cause: USER_NOT_REGISTERED" when extension registration is present


Hi,

I've noticed what appears to be a bug which I wanted to bring to your
attention.  Sometimes when FS is started, extensions can't be called even
after they've registered.  This doesn't happen all the time. and restarting
FS always solves the problem.  Just to be clear - this problem only
sometimes occurs when FS has been started/restarted, like after an update.
Once FS has "started successfully" and sees presence for registered
extensions, it continues to work fine.

It's all or nothing - no extensions can be dialed when this issue is
present, whether they're local or NATted.  If one extension can be reached,
you can reach them all (if they're registered, of course).

See debug info below on an attempt to dial a local (no NAT) extension - this
was one of the recent beta git pulls and the problem has been present
through several updates.  Let me know if you need any additional info.


freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostn
ame
102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;tra
nsport=udp,1333338853,192.168.1.4,5060,udp,server.home
257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.xx.xx:5060
;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;t
ransport=udp,1333338743,192.168.1.4,5060,udp,server.home
227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.xx.xx:1066;tra
nsport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.xx.xx:1068;tra
nsport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.xx.xx:1067;tra
nsport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.xx.xx:54629
;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.xx.xx:6480;
transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.xx
.xx:1062,1333338779,74.93.xx.xx,1062,udp,server.home

9 total.

2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
acl "domains". Falling back to Digest auth.
2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
(INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
192.168.1.4
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
acl "domains". Falling back to Digest auth.
2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
v=0
o=MxSIP 0 1 IN IP4 192.168.1.4
s=SIP Call
c=IN IP4 192.168.1.4
t=0 0
m=audio 16384 RTP/AVP 0 110 8 9
a=rtpmap:0 PCMU/8000
a=rtpmap:110 PCMU/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=silenceSupp:off - - - -
a=ptime:20

2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
(sofia/internal/102 at 192.168.1.6:5060) State INIT
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85
sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
(sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
(sofia/internal/102 at 192.168.1.6:5060) State ROUTING
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148
sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
<102>->2001 in context default
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call_direction] continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_direction]
${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
set(call_direction=local)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tod_example]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
[tod_example] break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->holiday_example] continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
[holiday_example] break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group-intercept] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [group-intercept]
destination_number(2001) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
break=never
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe
r})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-2]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
destination_number(2001) =~ /^\*9001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-1]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
destination_number(2001) =~ /^\*9000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call_privacy] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->call_return]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->del-group]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->add-group]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call-group-simo] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call-group-simo]
destination_number(2001) =~ /^\*\*82(\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->call-group-order] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->intercept-ext] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [intercept-ext]
destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Local_Extension_Skinny] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->send_to_voicemail] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.10d]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
destination_number(2001) =~ /^9(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.11d]
continue=true
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
destination_number(2001) =~ /^9(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->SipBroker.SIPDial] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->SipBroker.Toll_Free] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[SipBroker.Toll_Free] destination_number(2001) =~
/^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->TollFreeGateway] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [TollFreeGateway]
destination_number(2001) =~ /^91?(8(00|55|66|77|88)[2-9]\d{6})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->FlowRoute.91d10or11] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7519.9.10d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
caller_id_number(102) =~ /519/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7519.9.11d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
caller_id_number(102) =~ /519/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7612.9.10d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
${toll_allow}(local,domestic) =~ /local/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
destination_number(2001) =~ /^9(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->HL7612.9.11d] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
${toll_allow}(local,domestic) =~ /local/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
destination_number(2001) =~ /^9(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
destination_number(2001) =~ /^7(\d{7})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d10]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
destination_number(2001) =~ /^8(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d11]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
destination_number(2001) =~ /^8(\d{11})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Time] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Time] destination_number(2001) =~ /^9170$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Date] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Date] destination_number(2001) =~ /^9171$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
Clock Date and Time] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Recordings]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
destination_number(2001) =~ /^\*(732)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Company_Directory] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Hersch_Lauren_Main] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Holiday_Announcement] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
admin phones.park] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
admin phones] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
phones] destination_number(2001) =~ /^1001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_sales] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_support] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->group_dial_billing] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
destination_number(2001) =~ /^operator$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
destination_number(2001) =~ /^vmain$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
destination_number(2001) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->nb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [nb_conferences]
destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->wb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wb_conferences]
destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->uwb_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [uwb_conferences]
destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->cdquality_conferences] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(2001) =~
/^\*9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
destination_number(2001) =~ /^\*5000$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->dynamic_conference] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->rtp_multicast_page] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->parking_slots] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [parking_slots]
destination_number(2001) =~ /^(\*59[0-9][0-9])$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->valet_park_in] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_in]
destination_number(2001) =~ /^\*(6000)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->valet_park_out] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_out]
destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
destination_number(2001) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
destination_number(2001) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
destination_number(2001) =~ /callpark/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
destination_number(2001) =~ /pickup/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
destination_number(2001) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->fax_receive]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
destination_number(2001) =~ /^\*9178$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->fax_transmit] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
destination_number(2001) =~ /^\*9179$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_180] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
destination_number(2001) =~ /^\*9180$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_183_uk_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_183_music_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->ringback_post_answer_music] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
destination_number(2001) =~ /^\*9191$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->show_info]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
destination_number(2001) =~ /^\*9192$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->video_record] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
destination_number(2001) =~ /^\*9193$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->video_playback] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_playback]
destination_number(2001) =~ /^\*9194$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->delay_echo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
destination_number(2001) =~ /^\*9195$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
destination_number(2001) =~ /^\*9196$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->milliwatt]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
destination_number(2001) =~ /^\*9197$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tone_stream]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
destination_number(2001) =~ /^\*9198$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->zrtp_enrollement] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
[zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->hold_music]
continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
destination_number(2001) =~ /^\*9664$/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
[default->Local_Extension] continue=false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [Local_Extension]
destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(dialed_extension=2001)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
export(dialed_extension=2001)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db ${domain_name}
2001 ${limit_max} ${limit_destination})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
execute_extension::dx XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s
trftime(%b)}/${strftime(%d)}/${uuid}.wav)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
execute_extension::att_xfer XML features)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
4000, 440.0, 480.0))
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(hangup_after_bridge=true)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(continue_on_fail=true)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe
r})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
bridge(user/${user_data(${destination_number}@${domain_name} attr
id)}@${domain_name})
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
${domain_name} ${dialed_extension})
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> CS_EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
(sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241
sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial/102/2001)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c
1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
2012 23:50:41 -0400)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012
23:50:41 -0400]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
(export_vars) [dialed_extension]=[2001]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
192.168.1.6_2001 max:5, interval:0
2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001 is
now 1/5
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
execute_extension::dx XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
execute_extension::dx XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bd
e-116c-4100-b269-a6be7455c6c1.wav)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bd
e-116c-4100-b269-a6be7455c6c1.wav
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
execute_extension::cf XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
execute_extension::cf XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
execute_extension::att_xfer XML features)
2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
execute_extension::att_xfer XML features
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
440.0, 480.0))
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0,
480.0)]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
4000, 440.0, 480.0))
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000,
440.0, 480.0)]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-call_return/2001/102)
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c
6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/internal/102 at 192.168.1.6:5060
hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
[dialed_extension]=[2001] to event
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
[dialed_extension]=[2001] to event
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
[sip_invite_domain]=[192.168.1.6]
2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
[presence_id]=[2001 at 192.168.1.6]
2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
Cause: USER_NOT_REGISTERED
EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111
sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
Disable 2833 dtmf and switch to INFO
2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
[sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 192.168.1.4
port 16384 codec: 0 ms: 20
2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms
2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
sofia/internal/102 at 192.168.1.6:5060!
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
sofia/internal/102 at 192.168.1.6:5060:
v=0
o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
s=FreeSWITCH
c=IN IP4 192.168.1.6
t=0 0
m=audio 23164 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
[sofia/internal/102 at 192.168.1.6:5060] has been answered
2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel
sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
confirmed.
EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
2001)
2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
Activated L16 at 8000hz 1 channels 20ms
2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup
sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
sofia/internal/102 at 192.168.1.6:5060 [KILL]
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
file
/usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.
wav
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
sofia/internal/102 at 192.168.1.6:5060 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/102 at 192.168.1.6:5060) State HANGUP
2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel
sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: NORMAL_CLEARING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> CS_REPORTING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/102 at 192.168.1.6:5060) State REPORTING
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
NORMAL_CLEARING
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
(sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
CS_DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/102 at 192.168.1.6:5060 [BREAK]
2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
(sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
(sofia/internal/102 at 192.168.1.6:5060) Ended
2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close Channel
sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
(sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/102 at 192.168.1.6:5060) State DESTROY
2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374
sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
freeswitch at internal> /bye
[root at server bin]#

Regards,

Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.com

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From mario_fs at mgtech.com  Tue Apr  3 00:13:39 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 13:13:39 -0700
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
	TIME_WAIT problems
Message-ID: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>

If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
Mario G

From anthony.minessale at gmail.com  Tue Apr  3 00:21:09 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 15:21:09 -0500
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <020701cd1109$037d8cf0$0a78a6d0$@com>
References: <020701cd1109$037d8cf0$0a78a6d0$@com>
Message-ID: <CAKbxfG_5KGqQOfnxmDu1gYn_iw_FxW6R8BMjepLU+phV51bcAQ@mail.gmail.com>

I will be happy to investigate once you have some logs.

Presence has nothing to do with completing calls.
the "show registrations" is not relevant for you, you need the "sofia
status profile internal reg"

To reiterate:
1) please move this to JIRA
2) please do not paste large blocks of text into emails into this list.

also if you can get it in the state that is not working try manually
running sofia_contact <user>@<host> from the cli
also try getting a gcore of the running process and do thread apply all bt


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From acrow at integrafin.co.uk  Tue Apr  3 00:24:36 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 02 Apr 2012 21:24:36 +0100
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <020701cd1109$037d8cf0$0a78a6d0$@com>
References: <020701cd1109$037d8cf0$0a78a6d0$@com>
Message-ID: <4F7A0B04.2070405@integrafin.co.uk>

Hi,

I pulled from git today and am seeing something similar, will try to get 
a trace and submit a JIRA bug. It was not happening before my pull (last 
version was pulled in the middle of Feb).

I have two profiles for extensions (internal and nat) and it seems to 
occur most often when trying to call from internal to nat registered 
devices.

Cheers

Alex

On 02/04/12 20:44, Phil Quesinberry wrote:
>
> The phone never stops working.  The problem is either present from the 
> time FS is started or it isn't.  I'll have tobring it down andtry to 
> get it to fail againafter hours so I can provide more detailed info. 
> To me, it appears from the log I capturedthat presence_id is 
> notalwaysbeing initialized properlywhen FS is startedbut I haven't 
> looked at the source code to see what's really happening.  I do have 
> the two FS books on the way though, which should give me a deeper 
> understanding of how things work.  The wiki only goes so far.
>
> Also, when I do a sofia status profile internal reg, I see a bunch 
> oflong expiredregistration entries.  Not sure if that couldsomehowbe 
> related to this problem.
>
> freeswitch at internal> sofia status profile internal reg
>
> Registrations:
>
> =================================================================================================
>
> Call-ID:        49e008f79bd41460
>
> User:           102 at 192.168.1.6
>
> Contact:        "Phil" <sip:102 at 192.168.1.4:5060;transport=udp>
>
> Agent:          Aastra 9480i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:39:06) 
> EXPSECS(271)
>
> Host:           Server
>
> IP:             192.168.1.4
>
> Port:           5060
>
> Auth-User:      102
>
> Auth-Realm:     192.168.1.6
>
> MWI-Account:    102 at 192.168.1.6
>
> Call-ID:        5f92c72e93a50f67
>
> User:           226 at 192.168.1.6
>
> Contact:        "Sonny" <sip:226 at 74.93.xx.xx:5060;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:49:47) 
> EXPSECS(-2681088)
>
> Host:           Server
>
> IP: 74.93.xx.xx
>
> Port:           1066
>
> Auth-User:      226
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    226 at 192.168.1.6
>
> Call-ID:        09856e1e62e97740
>
> User:           227 at 192.168.1.6
>
> Contact:        "Laura" <sip:227 at 74.93.xx.xx:1064;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:47:39) 
> EXPSECS(-2681216)
>
> Host:           Server
>
> IP: 74.93.xx.xx
>
> Port:           5060
>
> Auth-User:      227
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    227 at 192.168.1.6
>
> Call-ID:        69748b87cb4b8d1d
>
> User:           257 at 192.168.1.6
>
> Contact:        "Tina" <sip:257 at 74.93.xx.xx:1060;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-02 13:49:11) 
> EXPSECS(-2681124)
>
> Host:           Server
>
> IP: 74.93.xx.xx
>
> Port:           1064
>
> Auth-User:      257
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    257 at 192.168.1.6
>
> Call-ID:        45249b30-85ddd13a at 10.0.0.159
>
> User:           519 at 192.168.1.6
>
> Contact:        Fax Machine <sip:519 at 74.93.xx.xx:1062>
>
> Agent:          Linksys/SPA2102-5.2.12
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:47) 
> EXPSECS(132)
>
> Host:           Server
>
> IP: 74.93.xx.xx
>
> Port:           1062
>
> Auth-User:      519
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    519 at 192.168.1.6
>
> Call-ID:        effe7aec41953215
>
> User:           102 at 192.168.2.105
>
> Contact:        "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
>
> Agent:          Aastra 9480i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-09 00:22:38) 
> EXPSECS(-2124717)
>
> Host:           localhost.localdomain
>
> IP:             192.168.2.104
>
> Port:           5060
>
> Auth-User:      102
>
> Auth-Realm:     192.168.2.105
>
> MWI-Account:    102 at 192.168.2.105
>
> Call-ID:        e9351a6c40675c86
>
> User:           102 at 192.168.2.6
>
> Contact:        "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
>
> Agent:          Aastra 9480i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-15 00:46:48) 
> EXPSECS(-1608467)
>
> Host:           pbx
>
> IP:             192.168.2.104
>
> Port:           5060
>
> Auth-User:      102
>
> Auth-Realm:     192.168.2.6
>
> MWI-Account:    102 at 192.168.2.6
>
> Call-ID:        45249b30-85ddd13a at 10.0.0.159
>
> User:           519 at 192.168.2.6
>
> Contact:        Fax Machine <sip:519 at 74.93.xx.xx:1062>
>
> Agent:          Linksys/SPA2102-5.2.12
>
> Status:         Registered(UDP)(unknown) EXP(2012-03-15 00:46:23) 
> EXPSECS(-1608492)
>
> Host:           pbx
>
> IP: 74.93.xx.xx
>
> Port:           1062
>
> Auth-User:      519
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    519 at 192.168.2.6
>
> Call-ID:        864e7aef4c59c423
>
> User:           225 at 192.168.1.6
>
> Contact:        "Vicky" <sip:225 at 74.93.xx.xx:1067;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:25) 
> EXPSECS(110)
>
> Host:           server
>
> IP: 74.93.xx.xx
>
> Port:           1067
>
> Auth-User:      225
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    225 at 192.168.1.6
>
> Call-ID:        29b0d19d8063e55c
>
> User:           226 at 192.168.1.6
>
> Contact:        "Sonny" <sip:226 at 74.93.xx.xx:1068;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:25) 
> EXPSECS(110)
>
> Host:           server
>
> IP: 74.93.xx.xx
>
> Port:           1068
>
> Auth-User:      226
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    226 at 192.168.1.6
>
> Call-ID:        6bc00b660278e069
>
> User:           227 at 192.168.1.6
>
> Contact:        "Laura" <sip:227 at 74.93.xx.xx:1066;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:35:51) 
> EXPSECS(76)
>
> Host:           server
>
> IP: 74.93.xx.xx
>
> Port:           1066
>
> Auth-User:      227
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    227 at 192.168.1.6
>
> Call-ID:        e721049935e82279
>
> User:           257 at 192.168.1.6
>
> Contact:        "Tina" <sip:257 at 74.93.xx.xx:5060;transport=udp>
>
> Agent:          Aastra 9143i/3.2.2.1136
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:32) 
> EXPSECS(117)
>
> Host:           server
>
> IP: 74.93.xx.xx
>
> Port:           5060
>
> Auth-User:      257
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    257 at 192.168.1.6
>
> Call-ID:        54667b49ca74cabe
>
> User:           3900 at 192.168.1.6
>
> Contact:        "Brad" 
> <sip:3900 at 173.66.xx.xx:54629;transport=udp;fs_nat=yes>
>
> Agent:          Aastra 9143i/3.2.2.2044
>
> Status:         Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:22) 
> EXPSECS(107)
>
> Host:           server.home
>
> IP: 173.66.xx.xx
>
> Port:           54629
>
> Auth-User:      3900
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    3900 at 192.168.1.6
>
> Call-ID:        6dc54f7994ffb61a
>
> User:           3901 at 192.168.1.6
>
> Contact:        "KCC Main" 
> <sip:3901 at 173.66.xx.xx:6480;transport=udp;fs_nat=yes>
>
> Agent:          Aastra 9143i/3.2.2.2044
>
> Status:         Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:38) 
> EXPSECS(123)
>
> Host:           server.home
>
> IP: 173.66.xx.xx
>
> Port:           6480
>
> Auth-User:      3901
>
> Auth-Realm:     qsystemseng.ourdomain.com
>
> MWI-Account:    3901 at 192.168.1.6
>
> Call-ID:        bae1af9975ebfa50
>
> User:           2001 at 192.168.1.6
>
> Contact:        "Phil" <sip:2001 at 192.168.1.4:5060;transport=udp>
>
> Agent:          Aastra 9480i/3.2.2.2044
>
> Status:         Registered(UDP)(unknown) EXP(2012-04-02 15:36:37) 
> EXPSECS(122)
>
> Host:           server.home
>
> IP:             192.168.1.4
>
> Port:           5060
>
> Auth-User:      2001
>
> Auth-Realm:     192.168.1.6
>
> MWI-Account:    2001 at 192.168.1.6
>
> Total items returned: 15
>
> =================================================================================================
>
> - Phil
>
> ----------
>
> Anthony Minessale wrote:
>
> plus the log is not detailed enough
>
> sofia global siptrace on
>
> sofia loglevel all 9
>
> console loglevel debug (/log debug)
>
> you would need enough log to cover the time when the phone is working
>
> until when it stops working.
>
> _____________________________________________
> *From:*Phil Quesinberry [mailto:philq at qsystemsengineering.com]
> *Sent:*Monday, April 02, 2012 12:37 AM
> *To:*'freeswitch-users at lists.freeswitch.org'
> *Subject:*Possible registration/presence bug - Sometimes get 
> "Originate Failed. Cause: USER_NOT_REGISTERED" when extension 
> registration is present
>
> Hi,
>
> I've noticed what appears to be a bug which I wanted to bring to your 
> attention.  Sometimes when FS is started, extensions can't be called 
> even after they've registered.  This doesn't happen all the time... 
> and restarting FS always solves the problem.  Just to be clear - this 
> problem only sometimes occurs when FS has been started/restarted, like 
> after an update.  Once FS has "started successfully" and sees presence 
> for registered extensions, it continues to work fine.
>
> It's all or nothing -- no extensions can be dialed when this issue is 
> present, whether they're local or NATted.  If one extension can be 
> reached, you can reach them all (if they're registered, of course).
>
> See debug info below on an attempt to dial a local (no NAT) extension 
> -- this was one of the recent beta git pulls and the problem has been 
> present through several updates.  Let me know if you need any 
> additional info.
>
>
> freeswitch at internal> show registrations
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>
> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>
> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.xx.xx:5060;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>
> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>
> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.xx.xx:1066;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>
> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.xx.xx:1068;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>
> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.xx.xx:1067;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>
> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.xx.xx:54629;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>
> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.xx.xx:6480;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>
> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.xx.xx:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>
> 9 total.
>
> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 
> Rejected by acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth 
> challenge (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] 
> from ip 192.168.1.4
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 
> Rejected by acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel 
> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel 
> sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>
> v=0
>
> o=MxSIP 0 1 IN IP4 192.168.1.4
>
> s=SIP Call
>
> c=IN IP4 192.168.1.4
>
> t=0 0
>
> m=audio 16384 RTP/AVP 0 110 8 9
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:110 PCMU/16000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:9 G722/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746 
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401 
> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85 
> sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125 
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401 
> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886 
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410 
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148 
> sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104 
> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>
> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing 
> Phil <102>->2001 in context default
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->unloop] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop] 
> ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop] 
> ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->call_direction] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action 
> set(call_direction=local)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->tod_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL) 
> [tod_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->holiday_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL) 
> [holiday_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->group-intercept] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [group-intercept] destination_number(2001) =~ /^\*8$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->redial] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial] 
> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->global] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global] 
> ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global] 
> ${sip_has_crypto}() =~ 
> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-last_dial/global/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->snom-demo-2] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [snom-demo-2] destination_number(2001) =~ /^\*9001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->snom-demo-1] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [snom-demo-1] destination_number(2001) =~ /^\*9000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->call_privacy] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [call_privacy] destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->call_return] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [call_return] destination_number(2001) =~ /^\*69$|^869$|^lcr$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->del-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group] 
> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->add-group] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group] 
> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->call-group-simo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [call-group-simo] destination_number(2001) =~ /^\*\*82(\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->call-group-order] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->intercept-ext] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [intercept-ext] destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Local_Extension_Skinny] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->send_to_voicemail] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->enum.9.10d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [enum.9.10d] destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->enum.9.11d] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [enum.9.11d] destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->SipBroker.SIPDial] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->SipBroker.Toll_Free] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [SipBroker.Toll_Free] destination_number(2001) =~ 
> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->TollFreeGateway] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [TollFreeGateway] destination_number(2001) =~ 
> /^91?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->FlowRoute.91d10or11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->HL7519.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [HL7519.9.10d] caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->HL7519.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [HL7519.9.11d] caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->HL7612.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) 
> [HL7612.9.10d] ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [HL7612.9.10d] destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->HL7612.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) 
> [HL7612.9.11d] ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [HL7612.9.11d] destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->enum.7d7] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7] 
> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->xmpp.8d10] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10] 
> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->xmpp.8d11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11] 
> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Talking Clock Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking 
> Clock Time] destination_number(2001) =~ /^9170$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Talking Clock Date] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking 
> Clock Date] destination_number(2001) =~ /^9171$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Talking Clock Date and Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking 
> Clock Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Recordings] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [Recordings] destination_number(2001) =~ /^\*(732)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Company_Directory] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Hersch_Lauren_Main] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Holiday_Announcement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring 
> all admin phones.park] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all 
> admin phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring 
> all admin phones] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all 
> admin phones] destination_number(2001) =~ /^1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->group_dial_sales] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->group_dial_support] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->group_dial_billing] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->operator] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator] 
> destination_number(2001) =~ /^operator$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain] 
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain] 
> destination_number(2001) =~ /^vmain$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->vmain1] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1] 
> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->vmain2] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2] 
> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->sip_uri] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri] 
> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->nb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [nb_conferences] destination_number(2001) =~ /^\*(30\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->wb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [wb_conferences] destination_number(2001) =~ /^\*(31\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->uwb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [uwb_conferences] destination_number(2001) =~ /^\*(32\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->cdquality_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->freeswitch_public_conf_via_sip] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [freeswitch_public_conf_via_sip] destination_number(2001) =~ 
> /^\*9(888|8888|1616|3232)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ivr_demo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo] 
> destination_number(2001) =~ /^\*5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->dynamic_conference] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->rtp_multicast_page] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [rtp_multicast_page] destination_number(2001) =~ 
> /^pagegroup$|^\*7243$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->parking_slots] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [parking_slots] destination_number(2001) =~ /^(\*59[0-9][0-9])$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->valet_park_in] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [valet_park_in] destination_number(2001) =~ /^\*(6000)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->valet_park_out] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [valet_park_out] destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park] 
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park] 
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park] 
> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->unpark] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark] 
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark] 
> destination_number(2001) =~ /^parking$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park] 
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park] 
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park] 
> destination_number(2001) =~ /callpark/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->unpark] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark] 
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark] 
> destination_number(2001) =~ /pickup/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait] 
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait] 
> destination_number(2001) =~ /^wait$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->fax_receive] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [fax_receive] destination_number(2001) =~ /^\*9178$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->fax_transmit] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [fax_transmit] destination_number(2001) =~ /^\*9179$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ringback_180] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [ringback_180] destination_number(2001) =~ /^\*9180$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ringback_183_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ringback_183_music_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ringback_post_answer_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ringback_post_answer_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/ 
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->ClueCon] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon] 
> destination_number(2001) =~ /^\*9191$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->show_info] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info] 
> destination_number(2001) =~ /^\*9192$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->video_record] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [video_record] destination_number(2001) =~ /^\*9193$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->video_playback] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->delay_echo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [delay_echo] destination_number(2001) =~ /^\*9195$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo] 
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo] 
> destination_number(2001) =~ /^\*9196$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->milliwatt] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt] 
> destination_number(2001) =~ /^\*9197$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->tone_stream] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [tone_stream] destination_number(2001) =~ /^\*9198$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->zrtp_enrollement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->hold_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) 
> [hold_music] destination_number(2001) =~ /^\*9664$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing 
> [default->Local_Extension] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) 
> [Local_Extension] destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> export(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db 
> ${domain_name} 2001 ${limit_max} ${limit_destination})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b 
> s execute_extension::dx XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b 
> s 
> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b 
> s execute_extension::cf XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b 
> s execute_extension::att_xfer XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(ringback=%(2000, 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(hangup_after_bridge=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(continue_on_fail=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} 
> var callgroup)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action 
> bridge(user/${user_data(${destination_number}@${domain_name} attr 
> id)}@${domain_name})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default 
> ${domain_name} ${dialed_extension})
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154 
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> 
> CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410 
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417 
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241 
> sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192 
> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-last_dial/102/2001)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 
> Apr 2012 23:50:41 -0400)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 
> 2012 23:50:41 -0400]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT 
> (export_vars) [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>
> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called: 
> 192.168.1.6_2001 max:5, interval:0
>
> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 
> 192.168.1.6_2001 is now 1/5
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s 
> execute_extension::dx XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: 
> *1 execute_extension::dx XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s 
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: 
> *2 
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s 
> execute_extension::cf XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: 
> *3 execute_extension::cf XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s 
> execute_extension::att_xfer XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: 
> *4 execute_extension::att_xfer XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000, 
> 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 
> 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 
> 4000, 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-call_return/2001/102)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281 
> sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 
> hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 
> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars] 
> [dialed_extension]=[2001] to event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing 
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047 
> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars] 
> [dialed_extension]=[2001] to event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing 
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing 
> variable [sip_invite_domain]=[192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing 
> variable [presence_id]=[2001 at 192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot 
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot 
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 
> Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate 
> Failed.  Cause: USER_NOT_REGISTERED
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec 
> Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec 
> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111 
> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.  
> Disable 2833 dtmf and switch to INFO
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP 
> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 
> 192.168.1.4 port 16384 codec: 0 ms: 20
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer 
> [soft] 160 bytes per 20ms
>
> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer 
> sofia/internal/102 at 192.168.1.6:5060!
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932 
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP 
> sofia/internal/102 at 192.168.1.6:5060:
>
> v=0
>
> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.1.6
>
> t=0 0
>
> m=audio 23164 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190 
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>
> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel 
> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel 
> sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel 
> sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
>
> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port 
> confirmed.
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 
> 192.168.1.6 2001)
>
> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec 
> Activated L16 at 8000hz 1 channels 20ms
>
> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848 
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>
> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup 
> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal 
> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done 
> playing file 
> /usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.wav
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287 
> sofia/internal/102 at 192.168.1.6:5060 skip receive message 
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417 
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602 
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel 
> sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47 
> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: 
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602 
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393 
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> 
> CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662 
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79 
> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause: 
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662 
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387 
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING -> 
> CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send 
> signal sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 
> 13 (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external 
> entities
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 
> 13 (sofia/internal/102 at 192.168.1.6:5060) Ended
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close 
> Channel sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491 
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494 
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504 
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374 
> sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86 
> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504 
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>
> freeswitch at internal> /bye
>
> [root at server bin]#
>
> Regards,
>
> /Phil Quesinberry/
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> _http://www.qsystemsengineering.com_
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
This message is intended only for the addressee and may contain
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From anthony.minessale at gmail.com  Tue Apr  3 00:28:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 15:28:15 -0500
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
 TIME_WAIT problems
In-Reply-To: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
Message-ID: <CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>

>From what I see, sofia uses the REUSEADDR flag so I will bet its yet
another MAC specific thing.
I am not really sure what you can do since you tend to find issues
deep inside libsofia-sip and its portability to OSX.
Maybe make your restart script sleep a bit?

libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
su_setreuseaddr(socket, 1);




On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
> Mario G
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mario_fs at mgtech.com  Tue Apr  3 00:33:08 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 2 Apr 2012 13:33:08 -0700
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
	TIME_WAIT problems
In-Reply-To: <CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
Message-ID: <E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>

That's what I was thinking, trying to figure out how now. Will post a solution when I get one.

On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:

>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
> another MAC specific thing.
> I am not really sure what you can do since you tend to find issues
> deep inside libsofia-sip and its portability to OSX.
> Maybe make your restart script sleep a bit?
> 
> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
> su_setreuseaddr(socket, 1);
> 
> 
> 
> 
> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
>> Mario G
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 



From abaci64 at gmail.com  Tue Apr  3 00:34:32 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 02 Apr 2012 16:34:32 -0400
Subject: [Freeswitch-users] context=any
Message-ID: <4F7A0D58.80705@gmail.com>

according to the wiki 
<http://wiki.freeswitch.org/wiki/Dialplan_XML#Context>the context name 
can be "any" in which case it would match any context. I tested it with 
context name="any" and it Doesn't seem to match so I wonder if this 
documentation is wrong or it's supposed to work. if someone can confirm 
to me that it's supposed to work that way I will file a Jira.
Thanks
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From wstephen80 at gmail.com  Tue Apr  3 00:38:55 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 2 Apr 2012 22:38:55 +0200
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se>
	<CAO4LeU08NY0r_Ztsm9SrOe2xajBikcmoH7WUWzo1AgRj0psmPg@mail.gmail.com>
	<CAKbxfG8mTk9v8KfP+ZBpgf8poVq7NV6HTvr-sxGEZbora_VLLw@mail.gmail.com>
	<CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
Message-ID: <CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>

Very satisfied with mod_posix_timer (with this I have solved the issue of
soft timer) but the problem of crash is present also with latest version.

I have open a jira:

http://jira.freeswitch.org/browse/FS-4064

Stephen

On Tue, Mar 27, 2012 at 10:52 PM, <cmrienzo at gmail.com> wrote:

> No big deal.  I fixed a few issues a while back, so maybe your problem is
> resolved.  Check the revision history to see if the changes happened after
> your crash.
>
>
>
>
> On Mar 27, 2012, at 14:24, Stephen Wilde <wstephen80 at gmail.com> wrote:
>
> The core dump was pretty unusable (stack corrupted) so I had some
> difficult to report to jira a useful info.
>
> On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>
>> Please report crashes in jira, otherwise they'll never get fixed.
>>
>>
>>
>> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I have tried to use mod_posix_timer but with this timer I had some FS
>>> crashes (segmentation fault) with core dump so I'm moved back to soft
>>> timers.
>>>
>>> Stephen
>>>
>>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:
>>>
>>>> You can also try mod_posix_timer if you have it built.  My CentOS 5.4
>>>> server performs better under load using it with heavy-timer vs using the
>>>> default timer.
>>>>
>>>> "timer_test 20 50 posix"
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> Ok, I'll try disabling the nanosleep.
>>>>>
>>>>> In server.2 the current load of the box is 35% and after FreeSwitch
>>>>> (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu).
>>>>>
>>>>>
>>>>> The result of "timer_test 20 50" is:
>>>>>
>>>>> http://pastebin.freeswitch.org/18756
>>>>>
>>>>>
>>>>> Here the result of "timer_test 10 50" :
>>>>>
>>>>> http://pastebin.freeswitch.org/18757
>>>>>
>>>>> It's normal that only the timer_test on 20 is bad where 10ms is good
>>>>> and also is good 40,60,120?:
>>>>>
>>>>> Stephen
>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Apr  3 01:02:06 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 16:02:06 -0500
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <4F7A0B04.2070405@integrafin.co.uk>
References: <020701cd1109$037d8cf0$0a78a6d0$@com>
	<4F7A0B04.2070405@integrafin.co.uk>
Message-ID: <CAKbxfG-x=e2ty2+6UYPpOWDqUdR6eRq7Sb-KDT-c-pcHYhOVdQ@mail.gmail.com>

I tracked this down to a race on startup where the core has not loaded
the ACL lists but sofia tries to use them for NAT detection.
I'm sure this is not something new but it should be fixed in HEAD.  (I
WISH SOMEONE WOULD HAVE OPENED A JIRA TO REFERENCE)


On Mon, Apr 2, 2012 at 3:24 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
> Hi,
>
> I pulled from git today and am seeing something similar, will try to get a
> trace and submit a JIRA bug. It was not happening before my pull (last
> version was pulled in the middle of Feb).
>
> I have two profiles for extensions (internal and nat) and it seems to occur
> most often when trying to call from internal to nat registered devices.
>
> Cheers
>
> Alex
>
> On 02/04/12 20:44, Phil Quesinberry wrote:
>
> The phone never stops working.? The problem is either present from the time
> FS is started or it isn?t.? I?ll have to bring it down and try to get it to
> fail again after hours so I can provide more detailed info.? To me, it
> appears from the log I captured that presence_id is not always being
> initialized properly when FS is started but I haven?t looked at the source
> code to see what?s really happening.? I do have the two FS books on the way
> though, which should give me a deeper understanding of how things work.? The
> wiki only goes so far.
>
> Also, when I do a sofia status profile internal reg, I see a bunch of long
> expired registration entries.? Not sure if that could somehow be related to
> this problem.
>
> freeswitch at internal> sofia status profile internal reg
>
> Registrations:
>
> =================================================================================================
>
> Call-ID:??????? 49e008f79bd41460
>
> User:?????????? 102 at 192.168.1.6
>
> Contact:??????? "Phil" <sip:102 at 192.168.1.4:5060;transport=udp>
>
> Agent:????????? Aastra 9480i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:39:06)
> EXPSECS(271)
>
> Host:?????????? Server
>
> IP:???????????? 192.168.1.4
>
> Port:?????????? 5060
>
> Auth-User:????? 102
>
> Auth-Realm:???? 192.168.1.6
>
> MWI-Account:??? 102 at 192.168.1.6
>
> Call-ID:??????? 5f92c72e93a50f67
>
> User:?????????? 226 at 192.168.1.6
>
> Contact:??????? "Sonny" <sip:226 at 74.93.xx.xx:5060;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-02 13:49:47)
> EXPSECS(-2681088)
>
> Host:?????????? Server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1066
>
> Auth-User:????? 226
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 226 at 192.168.1.6
>
> Call-ID:??????? 09856e1e62e97740
>
> User:?????????? 227 at 192.168.1.6
>
> Contact:??????? "Laura" <sip:227 at 74.93.xx.xx:1064;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-02 13:47:39)
> EXPSECS(-2681216)
>
> Host:?????????? Server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 5060
>
> Auth-User:????? 227
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 227 at 192.168.1.6
>
> Call-ID:??????? 69748b87cb4b8d1d
>
> User:?????????? 257 at 192.168.1.6
>
> Contact:??????? "Tina" <sip:257 at 74.93.xx.xx:1060;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-02 13:49:11)
> EXPSECS(-2681124)
>
> Host:?????????? Server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1064
>
> Auth-User:????? 257
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 257 at 192.168.1.6
>
> Call-ID:??????? 45249b30-85ddd13a at 10.0.0.159
>
> User:?????????? 519 at 192.168.1.6
>
> Contact:??????? Fax Machine <sip:519 at 74.93.xx.xx:1062>
>
> Agent:????????? Linksys/SPA2102-5.2.12
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:36:47)
> EXPSECS(132)
>
> Host:?????????? Server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1062
>
> Auth-User:????? 519
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 519 at 192.168.1.6
>
> Call-ID:??????? effe7aec41953215
>
> User:?????????? 102 at 192.168.2.105
>
> Contact:??????? "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
>
> Agent:????????? Aastra 9480i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-09 00:22:38)
> EXPSECS(-2124717)
>
> Host:?????????? localhost.localdomain
>
> IP:???????????? 192.168.2.104
>
> Port:?????????? 5060
>
> Auth-User:????? 102
>
> Auth-Realm:???? 192.168.2.105
>
> MWI-Account:??? 102 at 192.168.2.105
>
> Call-ID:??????? e9351a6c40675c86
>
> User:?????????? 102 at 192.168.2.6
>
> Contact:??????? "Phil" <sip:102 at 192.168.2.104:5060;transport=udp>
>
> Agent:????????? Aastra 9480i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-15 00:46:48)
> EXPSECS(-1608467)
>
> Host:?????????? pbx
>
> IP:???????????? 192.168.2.104
>
> Port:?????????? 5060
>
> Auth-User:????? 102
>
> Auth-Realm:???? 192.168.2.6
>
> MWI-Account:??? 102 at 192.168.2.6
>
> Call-ID:??????? 45249b30-85ddd13a at 10.0.0.159
>
> User:?????????? 519 at 192.168.2.6
>
> Contact:??????? Fax Machine <sip:519 at 74.93.xx.xx:1062>
>
> Agent:????????? Linksys/SPA2102-5.2.12
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-03-15 00:46:23)
> EXPSECS(-1608492)
>
> Host:?????????? pbx
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1062
>
> Auth-User:????? 519
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 519 at 192.168.2.6
>
> Call-ID:??????? 864e7aef4c59c423
>
> User:?????????? 225 at 192.168.1.6
>
> Contact:??????? "Vicky" <sip:225 at 74.93.xx.xx:1067;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:36:25)
> EXPSECS(110)
>
> Host:?????????? server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1067
>
> Auth-User:????? 225
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 225 at 192.168.1.6
>
> Call-ID:??????? 29b0d19d8063e55c
>
> User:?????????? 226 at 192.168.1.6
>
> Contact:??????? "Sonny" <sip:226 at 74.93.xx.xx:1068;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:36:25)
> EXPSECS(110)
>
> Host:?????????? server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1068
>
> Auth-User:????? 226
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 226 at 192.168.1.6
>
> Call-ID:??????? 6bc00b660278e069
>
> User:?????????? 227 at 192.168.1.6
>
> Contact:??????? "Laura" <sip:227 at 74.93.xx.xx:1066;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:35:51)
> EXPSECS(76)
>
> Host:?????????? server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 1066
>
> Auth-User:????? 227
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 227 at 192.168.1.6
>
> Call-ID:??????? e721049935e82279
>
> User:?????????? 257 at 192.168.1.6
>
> Contact:??????? "Tina" <sip:257 at 74.93.xx.xx:5060;transport=udp>
>
> Agent:????????? Aastra 9143i/3.2.2.1136
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:36:32)
> EXPSECS(117)
>
> Host:?????????? server
>
> IP:???????????? 74.93.xx.xx
>
> Port:?????????? 5060
>
> Auth-User:????? 257
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 257 at 192.168.1.6
>
> Call-ID:??????? 54667b49ca74cabe
>
> User:?????????? 3900 at 192.168.1.6
>
> Contact:??????? "Brad"
> <sip:3900 at 173.66.xx.xx:54629;transport=udp;fs_nat=yes>
>
> Agent:????????? Aastra 9143i/3.2.2.2044
>
> Status:???????? Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:22)
> EXPSECS(107)
>
> Host:?????????? server.home
>
> IP:???????????? 173.66.xx.xx
>
> Port:?????????? 54629
>
> Auth-User:????? 3900
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 3900 at 192.168.1.6
>
> Call-ID:??????? 6dc54f7994ffb61a
>
> User:?????????? 3901 at 192.168.1.6
>
> Contact:??????? "KCC Main"
> <sip:3901 at 173.66.xx.xx:6480;transport=udp;fs_nat=yes>
>
> Agent:????????? Aastra 9143i/3.2.2.2044
>
> Status:???????? Registered(AUTO-NAT)(unknown) EXP(2012-04-02 15:36:38)
> EXPSECS(123)
>
> Host:?????????? server.home
>
> IP:???????????? 173.66.xx.xx
>
> Port:?????????? 6480
>
> Auth-User:????? 3901
>
> Auth-Realm:???? qsystemseng.ourdomain.com
>
> MWI-Account:??? 3901 at 192.168.1.6
>
> Call-ID:??????? bae1af9975ebfa50
>
> User:?????????? 2001 at 192.168.1.6
>
> Contact:??????? "Phil" <sip:2001 at 192.168.1.4:5060;transport=udp>
>
> Agent:????????? Aastra 9480i/3.2.2.2044
>
> Status:???????? Registered(UDP)(unknown) EXP(2012-04-02 15:36:37)
> EXPSECS(122)
>
> Host:?????????? server.home
>
> IP:???????????? 192.168.1.4
>
> Port:?????????? 5060
>
> Auth-User:????? 2001
>
> Auth-Realm:???? 192.168.1.6
>
> MWI-Account:??? 2001 at 192.168.1.6
>
> Total items returned: 15
>
> =================================================================================================
>
> - Phil
>
> ----------
>
> Anthony Minessale wrote:
>
> plus the log is not detailed enough
>
> sofia global siptrace on
>
> sofia loglevel all 9
>
> console loglevel debug (/log debug)
>
> you would need enough log to cover the time when the phone is working
>
> until when it stops working.
>
> _____________________________________________
> From: Phil Quesinberry [mailto:philq at qsystemsengineering.com]
> Sent: Monday, April 02, 2012 12:37 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: Possible registration/presence bug - Sometimes get "Originate
> Failed. Cause: USER_NOT_REGISTERED" when extension registration is present
>
> Hi,
>
> I?ve noticed what appears to be a bug which I wanted to bring to your
> attention.? Sometimes when FS is started, extensions can?t be called even
> after they?ve registered.? This doesn?t happen all the time? and restarting
> FS always solves the problem.? Just to be clear - this problem only
> sometimes occurs when FS has been started/restarted, like after an update.
> Once FS has ?started successfully? and sees presence for registered
> extensions, it continues to work fine.
>
> It?s all or nothing ? no extensions can be dialed when this issue is
> present, whether they?re local or NATted.? If one extension can be reached,
> you can reach them all (if they?re registered, of course).
>
> See debug info below on an attempt to dial a local (no NAT) extension ? this
> was one of the recent beta git pulls and the problem has been present
> through several updates.? Let me know if you need any additional info.
>
>
> freeswitch at internal> show registrations
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname
>
> 102,192.168.1.6,a7289b6393a8023e,sofia/internal/sip:102 at 192.168.1.4:5060;transport=udp,1333338853,192.168.1.4,5060,udp,server.home
>
> 257,192.168.1.6,aab02ec49af03567,sofia/internal/sip:257 at 74.93.222.xx.xx:5060;transport=udp,1333338709,74.93.xx.xx,5060,udp,server.home
>
> 2001,192.168.1.6,bae1af9975ebfa50,sofia/internal/sip:2001 at 192.168.1.4:5060;transport=udp,1333338743,192.168.1.4,5060,udp,server.home
>
> 227,192.168.1.6,42db21f84f52e3ef,sofia/internal/sip:227 at 74.93.xx.xx:1066;transport=udp,1333338747,74.93.xx.xx,1066,udp,server.home
>
> 226,192.168.1.6,25619841fc53c75e,sofia/internal/sip:226 at 74.93.xx.xx:1068;transport=udp,1333338751,74.93.xx.xx,1068,udp,server.home
>
> 225,192.168.1.6,308cce2ca1734cdc,sofia/internal/sip:225 at 74.93.xx.xx:1067;transport=udp,1333338751,74.93.xx.xx,1067,udp,server.home
>
> 3900,192.168.1.6,54667b49ca74cabe,sofia/internal/sip:3900 at 173.66.xx.xx:54629;transport=udp;fs_nat=yes,1333338762,173.66.xx.xx,54629,udp,server.home
>
> 3901,192.168.1.6,6dc54f7994ffb61a,sofia/internal/sip:3901 at 173.66.xx.xx:6480;transport=udp;fs_nat=yes,1333338779,173.66.xx.xx,6480,udp,server.home
>
> 519,192.168.1.6,45249b30-85ddd13a at 10.0.0.159,sofia/internal/sip:519 at 74.93.xx.xx:1062,1333338779,74.93.xx.xx,1062,udp,server.home
>
> 9 total.
>
> 2012-04-01 23:50:41.449612 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.449612 [WARNING] sofia_reg.c:1422 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [2001 at 192.168.1.6] from ip
> 192.168.1.4
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:7576 IP 192.168.1.4 Rejected by
> acl "domains". Falling back to Digest auth.
>
> 2012-04-01 23:50:41.549553 [NOTICE] switch_channel.c:926 New Channel
> sofia/internal/102 at 192.168.1.6:5060 [9a9a3bde-116c-4100-b269-a6be7455c6c1]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5541 Channel
> sofia/internal/102 at 192.168.1.6:5060 entering state [received][100]
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5552 Remote SDP:
>
> v=0
>
> o=MxSIP 0 1 IN IP4 192.168.1.4
>
> s=SIP Call
>
> c=IN IP4 192.168.1.4
>
> t=0 0
>
> m=audio 16384 RTP/AVP 0 110 8 9
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:110 PCMU/16000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:9 G722/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> 2012-04-01 23:50:41.549553 [DEBUG] sofia.c:5746
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_NEW -> CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:85
> sofia/internal/102 at 192.168.1.6:5060 SOFIA INIT
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:125
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_INIT -> CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:401
> (sofia/internal/102 at 192.168.1.6:5060) State INIT going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1886
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change DOWN -> RINGING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:148
> sofia/internal/102 at 192.168.1.6:5060 SOFIA ROUTING
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/102 at 192.168.1.6:5060 Standard ROUTING
>
> 2012-04-01 23:50:41.549553 [INFO] mod_dialplan_xml.c:485 Processing Phil
> <102>->2001 in context default
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unloop]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_direction] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_direction]
> ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 ANTI-Action
> set(call_direction=local)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tod_example]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [tod_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->holiday_example] continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Date/TimeMatch (FAIL)
> [holiday_example] break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group-intercept] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [group-intercept]
> destination_number(2001) =~ /^\*8$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->redial]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [redial]
> destination_number(2001) =~ /^(redial|\*870)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->global]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${call_debug}(false) =~ /^true$/ break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [global]
> ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Absolute Condition [global]
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/global/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-2]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-2]
> destination_number(2001) =~ /^\*9001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->snom-demo-1]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [snom-demo-1]
> destination_number(2001) =~ /^\*9000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call_privacy] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_privacy]
> destination_number(2001) =~ /^\*67(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->call_return]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call_return]
> destination_number(2001) =~ /^\*69$|^869$|^lcr$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->del-group]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [del-group]
> destination_number(2001) =~ /^\*\*80(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->add-group]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [add-group]
> destination_number(2001) =~ /^\*\*81(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-simo] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [call-group-simo]
> destination_number(2001) =~ /^\*\*82(\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->call-group-order] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [call-group-order] destination_number(2001) =~ /^\*83(\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->intercept-ext] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [intercept-ext]
> destination_number(2001) =~ /^\*\*(\d+)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension_Skinny] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Local_Extension_Skinny] destination_number(2001) =~ /^(11[01][0-9])$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->send_to_voicemail] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [send_to_voicemail] destination_number(2001) =~ /^\*99(\d{2,7})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.10d]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.9.11d]
> continue=true
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.SIPDial] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.SIPDial] destination_number(2001) =~ /^([*]\d{5,})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->SipBroker.Toll_Free] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [SipBroker.Toll_Free] destination_number(2001) =~
> /^91?(8(00|66|77|88)[2-9]\d{6})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->TollFreeGateway] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [TollFreeGateway]
> destination_number(2001) =~ /^91?(8(00|55|66|77|88)[2-9]\d{6})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->FlowRoute.91d10or11] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [FlowRoute.91d10or11] destination_number(2001) =~ /^91?(\d{10})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.10d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7519.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7519.9.11d]
> caller_id_number(102) =~ /519/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.10d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.10d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.10d]
> destination_number(2001) =~ /^9(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->HL7612.9.11d] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [HL7612.9.11d]
> ${toll_allow}(local,domestic) =~ /local/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [HL7612.9.11d]
> destination_number(2001) =~ /^9(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->enum.7d7]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [enum.7d7]
> destination_number(2001) =~ /^7(\d{7})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d10]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d10]
> destination_number(2001) =~ /^8(\d{10})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->xmpp.8d11]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [xmpp.8d11]
> destination_number(2001) =~ /^8(\d{11})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Time] destination_number(2001) =~ /^9170$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date] destination_number(2001) =~ /^9171$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Talking
> Clock Date and Time] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Talking Clock
> Date and Time] destination_number(2001) =~ /^9172$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Recordings]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Recordings]
> destination_number(2001) =~ /^\*(732)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Company_Directory] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Company_Directory] destination_number(2001) =~ /^5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Hersch_Lauren_Main] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Hersch_Lauren_Main] destination_number(2001) =~ /^5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Holiday_Announcement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [Holiday_Announcement] destination_number(2001) =~ /^5010$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones.park] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
> phones.park] destination_number(2001) =~ /^\*1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->Ring all
> admin phones] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [Ring all admin
> phones] destination_number(2001) =~ /^1001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_sales] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_sales] destination_number(2001) =~ /^\*2000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_support] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_support] destination_number(2001) =~ /^\*2001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->group_dial_billing] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [group_dial_billing] destination_number(2001) =~ /^\*2002$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->operator]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [operator]
> destination_number(2001) =~ /^operator$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain]
> destination_number(2001) =~ /^vmain$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain1]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain1]
> destination_number(2001) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->vmain2]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [vmain2]
> destination_number(2001) =~ /^vmain2$|^\*98$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->sip_uri]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [sip_uri]
> destination_number(2001) =~ /^sip:(.*)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->nb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [nb_conferences]
> destination_number(2001) =~ /^\*(30\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->wb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wb_conferences]
> destination_number(2001) =~ /^\*(31\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->uwb_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [uwb_conferences]
> destination_number(2001) =~ /^\*(32\d{2})$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->cdquality_conferences] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [cdquality_conferences] destination_number(2001) =~ /^\*(33\d{2})$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->freeswitch_public_conf_via_sip] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(2001) =~
> /^\*9(888|8888|1616|3232)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ivr_demo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ivr_demo]
> destination_number(2001) =~ /^\*5000$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->dynamic_conference] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [dynamic_conference] destination_number(2001) =~ /^\*5001$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->rtp_multicast_page] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [rtp_multicast_page] destination_number(2001) =~ /^pagegroup$|^\*7243$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->parking_slots] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [parking_slots]
> destination_number(2001) =~ /^(\*59[0-9][0-9])$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_in] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_in]
> destination_number(2001) =~ /^\*(6000)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->valet_park_out] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [valet_park_out]
> destination_number(2001) =~ /^\*(60\d\d)$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /park\+(\d+)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /^parking$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->park]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [park]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [park]
> destination_number(2001) =~ /callpark/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->unpark]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [unpark]
> source(mod_sofia) =~ /mod_sofia/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [unpark]
> destination_number(2001) =~ /pickup/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->wait]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [wait]
> destination_number(2001) =~ /^wait$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->fax_receive]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_receive]
> destination_number(2001) =~ /^\*9178$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->fax_transmit] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [fax_transmit]
> destination_number(2001) =~ /^\*9179$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_180] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ringback_180]
> destination_number(2001) =~ /^\*9180$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_uk_ring] destination_number(2001) =~ /^\*9181$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_183_music_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_183_music_ring] destination_number(2001) =~ /^\*9182$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_uk_ring] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(2001) =~ /^\*9183$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->ringback_post_answer_music] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [ringback_post_answer_music] destination_number(2001) =~ /^\*9184$/
> break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->ClueCon]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [ClueCon]
> destination_number(2001) =~ /^\*9191$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->show_info]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [show_info]
> destination_number(2001) =~ /^\*9192$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_record] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_record]
> destination_number(2001) =~ /^\*9193$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->video_playback] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [video_playback]
> destination_number(2001) =~ /^\*9194$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->delay_echo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [delay_echo]
> destination_number(2001) =~ /^\*9195$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->echo]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [echo]
> destination_number(2001) =~ /^\*9196$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->milliwatt]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [milliwatt]
> destination_number(2001) =~ /^\*9197$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->tone_stream]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [tone_stream]
> destination_number(2001) =~ /^\*9198$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->zrtp_enrollement] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL)
> [zrtp_enrollement] destination_number(2001) =~ /^\*9787$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing [default->hold_music]
> continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (FAIL) [hold_music]
> destination_number(2001) =~ /^\*9664$/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 parsing
> [default->Local_Extension] continue=false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Regex (PASS) [Local_Extension]
> destination_number(2001) =~ /(^\d{2,7}$)/ break=on-false
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> export(dialed_extension=2001)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action limit(db ${domain_name}
> 2001 ${limit_max} ${limit_destination})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(ringback=%(2000,
> 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(transfer_ringback=%(2000, 4000, 440.0, 480.0))
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action set(call_timeout=24)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(hangup_after_bridge=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(continue_on_fail=true)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
> var callgroup)})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action
> bridge(user/${user_data(${destination_number}@${domain_name} attr
> id)}@${domain_name})
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action answer()
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action sleep(1000)
>
> Dialplan: sofia/internal/102 at 192.168.1.6:5060 Action voicemail(default
> ${domain_name} ${dialed_extension})
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:154
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_ROUTING -> CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:410
> (sofia/internal/102 at 192.168.1.6:5060) State ROUTING going to sleep
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_sofia.c:241
> sofia/internal/102 at 192.168.1.6:5060 SOFIA EXECUTE
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_core_state_machine.c:192
> sofia/internal/102 at 192.168.1.6:5060 Standard EXECUTE
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_direction=local)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [call_direction]=[local]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-spymap/102/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-last_dial/102/2001)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-last_dial/global/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(RFC2822_DATE=Sun, 01 Apr
> 2012 23:50:41 -0400)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [RFC2822_DATE]=[Sun, 01 Apr 2012
> 23:50:41 -0400]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 export(dialed_extension=2001)
>
> 2012-04-01 23:50:41.549553 [DEBUG] switch_channel.c:1093 EXPORT
> (export_vars) [dialed_extension]=[2001]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 limit(db 192.168.1.6 2001 5 )
>
> 2012-04-01 23:50:41.549553 [INFO] switch_limit.c:126 incr called:
> 192.168.1.6_2001 max:5, interval:0
>
> 2012-04-01 23:50:41.569549 [INFO] mod_db.c:197 Usage for 192.168.1.6_2001 is
> now 1/5
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(1 b s
> execute_extension::dx XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *1
> execute_extension::dx XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(2 b s
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *2
> record_session::/usr/local/freeswitch/recordings/archive/2012/Apr/01/9a9a3bde-116c-4100-b269-a6be7455c6c1.wav
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(3 b s
> execute_extension::cf XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *3
> execute_extension::cf XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bind_meta_app(4 b s
> execute_extension::att_xfer XML features)
>
> 2012-04-01 23:50:41.569549 [INFO] switch_ivr_async.c:3204 Bound B-Leg: *4
> execute_extension::att_xfer XML features
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(ringback=%(2000, 4000,
> 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [ringback]=[%(2000, 4000, 440.0,
> 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(transfer_ringback=%(2000,
> 4000, 440.0, 480.0))
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [transfer_ringback]=[%(2000, 4000,
> 440.0, 480.0)]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(call_timeout=24)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [call_timeout]=[24]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(hangup_after_bridge=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [hangup_after_bridge]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-call_return/2001/102)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-last_dial_ext/2001/9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(called_party_callgroup=)
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_dptools.c:1281
> sofia/internal/102 at 192.168.1.6:5060 SET [called_party_callgroup]=[UNDEF]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060
> hash(insert/192.168.1.6-last_dial//9a9a3bde-116c-4100-b269-a6be7455c6c1)
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/2001 at 192.168.1.6)
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
> [dialed_extension]=[2001] to event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:1047
> sofia/internal/102 at 192.168.1.6:5060 EXPORTING[export_vars]
> [dialed_extension]=[2001] to event
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:1884 Parsing
> global variables
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [sip_invite_domain]=[192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_event.c:1522 Parsing variable
> [presence_id]=[2001 at 192.168.1.6]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [NOTICE] switch_ivr_originate.c:2459 Cannot
> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_ivr_originate.c:3364 Originate
> Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
>
> 2012-04-01 23:50:41.569549 [INFO] mod_dptools.c:2922 Originate Failed.
> Cause: USER_NOT_REGISTERED
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 answer()
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:4876 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:2996 Set Codec
> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_codec.c:111
> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:5008 No 2833 in SDP.
> Disable 2833 dtmf and switch to INFO
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia_glue.c:3248 AUDIO RTP
> [sofia/internal/102 at 192.168.1.6:5060] 192.168.1.6 port 23164 -> 192.168.1.4
> port 16384 codec: 0 ms: 20
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
>
> 2012-04-01 23:50:41.569549 [NOTICE] sofia_glue.c:4022 Pre-Answer
> sofia/internal/102 at 192.168.1.6:5060!
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:2932
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> EARLY
>
> 2012-04-01 23:50:41.569549 [DEBUG] mod_sofia.c:750 Local SDP
> sofia/internal/102 at 192.168.1.6:5060:
>
> v=0
>
> o=FreeSWITCH 1333315477 1333315478 IN IP4 192.168.1.6
>
> s=FreeSWITCH
>
> c=IN IP4 192.168.1.6
>
> t=0 0
>
> m=audio 23164 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=silenceSupp:off - - - -
>
> a=ptime:20
>
> a=sendrecv
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_channel.c:3190
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change EARLY -> ACTIVE
>
> 2012-04-01 23:50:41.569549 [NOTICE] mod_dptools.c:1135 Channel
> [sofia/internal/102 at 192.168.1.6:5060] has been answered
>
> 2012-04-01 23:50:41.569549 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.569549 [DEBUG] sofia.c:5541 Channel
> sofia/internal/102 at 192.168.1.6:5060 entering state [completed][200]
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 sleep(1000)
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.649589 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:41.669832 [DEBUG] sofia.c:5541 Channel
> sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200]
>
> 2012-04-01 23:50:41.849781 [DEBUG] switch_rtp.c:3224 Correct ip/port
> confirmed.
>
> EXECUTE sofia/internal/102 at 192.168.1.6:5060 voicemail(default 192.168.1.6
> 2001)
>
> 2012-04-01 23:50:42.689527 [DEBUG] switch_ivr_play_say.c:1306 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> 2012-04-01 23:50:46.249491 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2848
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change ACTIVE -> HANGUP
>
> 2012-04-01 23:50:46.269490 [NOTICE] sofia.c:628 Hangup
> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_channel.c:2871 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [KILL]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_ivr_play_say.c:1678 done playing
> file
> /usr/local/freeswitch/storage/voicemail/default/192.168.1.6/2001/greeting_1.wav
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:2287
> sofia/internal/102 at 192.168.1.6:5060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/102 at 192.168.1.6:5060) State EXECUTE going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP
>
> 2012-04-01 23:50:46.269490 [DEBUG] mod_sofia.c:469 Channel
> sofia/internal/102 at 192.168.1.6:5060 hanging up, cause: NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:47
> sofia/internal/102 at 192.168.1.6:5060 Standard HANGUP, cause: NORMAL_CLEARING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/102 at 192.168.1.6:5060) State HANGUP going to sleep
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:393
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_HANGUP -> CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_REPORTING
>
> 2012-04-01 23:50:46.269490 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:79
> sofia/internal/102 at 192.168.1.6:5060 Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/102 at 192.168.1.6:5060) State REPORTING going to sleep
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:387
> (sofia/internal/102 at 192.168.1.6:5060) State Change CS_REPORTING ->
> CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/102 at 192.168.1.6:5060 [BREAK]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_session.c:1382 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Locked, Waiting on external entities
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1400 Session 13
> (sofia/internal/102 at 192.168.1.6:5060) Ended
>
> 2012-04-01 23:50:46.289510 [NOTICE] switch_core_session.c:1402 Close Channel
> sofia/internal/102 at 192.168.1.6:5060 [CS_DESTROY]
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change HANGUP -> DOWN
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:494
> (sofia/internal/102 at 192.168.1.6:5060) Running State Change CS_DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] mod_sofia.c:374
> sofia/internal/102 at 192.168.1.6:5060 SOFIA DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:86
> sofia/internal/102 at 192.168.1.6:5060 Standard DESTROY
>
> 2012-04-01 23:50:46.289510 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/102 at 192.168.1.6:5060) State DESTROY going to sleep
>
> freeswitch at internal> /bye
>
> [root at server bin]#
>
> Regards,
>
> Phil Quesinberry
>
> Q Systems Engineering, Inc.
>
> Electronic Controls and Embedded Systems Development
>
> (410) 969-8002
>
> http://www.qsystemsengineering.com
>
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> This message is intended only for the addressee and may contain
> confidential information.  Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
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> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number:
> 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the
> FSA Register; number: 190856)
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>
> _________________________________________________________________________
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>
> 
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>
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Tue Apr  3 01:11:25 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Apr 2012 16:11:25 -0500
Subject: [Freeswitch-users] context=any
In-Reply-To: <4F7A0D58.80705@gmail.com>
References: <4F7A0D58.80705@gmail.com>
Message-ID: <CAKbxfG9a_LnoNkoCpp19AY2Ld9j7c=sEwRZUAv_pae8wP1cpMg@mail.gmail.com>

I believe that is incorrect.

You can set the context in sofia to "_domain_" which will set the
context dynamically to the value of the host in the From: header.


On Mon, Apr 2, 2012 at 3:34 PM, Abaci <abaci64 at gmail.com> wrote:
> according to the wiki the context name can be "any" in which case it would
> match any context. I tested it with context name="any" and it Doesn't seem
> to match so I wonder if this documentation is wrong or it's supposed to
> work. if someone can confirm to me that it's supposed to work that way I
> will file a Jira.
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Tue Apr  3 01:18:54 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 14:18:54 -0700
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <CAKbxfG-x=e2ty2+6UYPpOWDqUdR6eRq7Sb-KDT-c-pcHYhOVdQ@mail.gmail.com>
References: <020701cd1109$037d8cf0$0a78a6d0$@com>
	<4F7A0B04.2070405@integrafin.co.uk>
	<CAKbxfG-x=e2ty2+6UYPpOWDqUdR6eRq7Sb-KDT-c-pcHYhOVdQ@mail.gmail.com>
Message-ID: <CAKzWOxWUTPPqR56vw+_iV8_cEQROvE1U43eUEoUG2LHsRTUQHQ@mail.gmail.com>

On Mon, Apr 2, 2012 at 2:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I tracked this down to a race on startup where the core has not loaded
> the ACL lists but sofia tries to use them for NAT detection.
> I'm sure this is not something new but it should be fixed in HEAD.  (I
> WISH SOMEONE WOULD HAVE OPENED A JIRA TO REFERENCE)
>

http://freeswitch.org/please_file_a_jira.wav
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From msc at freeswitch.org  Tue Apr  3 01:25:47 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Apr 2012 14:25:47 -0700
Subject: [Freeswitch-users] context=any
In-Reply-To: <CAKbxfG9a_LnoNkoCpp19AY2Ld9j7c=sEwRZUAv_pae8wP1cpMg@mail.gmail.com>
References: <4F7A0D58.80705@gmail.com>
	<CAKbxfG9a_LnoNkoCpp19AY2Ld9j7c=sEwRZUAv_pae8wP1cpMg@mail.gmail.com>
Message-ID: <CAKzWOxUT+v9KKGyybQj8R_bq-Nj1Jmfb5X1+JXCUyTLGV14ESA@mail.gmail.com>

On Mon, Apr 2, 2012 at 2:11 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I believe that is incorrect.
>
> You can set the context in sofia to "_domain_" which will set the
> context dynamically to the value of the host in the From: header.
>

Yeah, I can't find any reference anywhere in the code base to "context"
ever being equal to "any". I specifically checked mod_dialplan_xml.c and
the only place in the code that uses the word "any" is in reference to the
"regex" tag. Abaci, please feel free to expunge that information from the
wiki. If you can test what Anthony mentioned and confirm that it works then
that would be a handy item to add to the wiki.

Thanks,
MC
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From abaci64 at gmail.com  Tue Apr  3 01:29:07 2012
From: abaci64 at gmail.com (Abaci)
Date: Mon, 02 Apr 2012 17:29:07 -0400
Subject: [Freeswitch-users] context=any
In-Reply-To: <CAKzWOxUT+v9KKGyybQj8R_bq-Nj1Jmfb5X1+JXCUyTLGV14ESA@mail.gmail.com>
References: <4F7A0D58.80705@gmail.com>
	<CAKbxfG9a_LnoNkoCpp19AY2Ld9j7c=sEwRZUAv_pae8wP1cpMg@mail.gmail.com>
	<CAKzWOxUT+v9KKGyybQj8R_bq-Nj1Jmfb5X1+JXCUyTLGV14ESA@mail.gmail.com>
Message-ID: <4F7A1A23.8090309@gmail.com>

Will do.

On 4/2/2012 5:25 PM, Michael Collins wrote:
>
>
> On Mon, Apr 2, 2012 at 2:11 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     I believe that is incorrect.
>
>     You can set the context in sofia to "_domain_" which will set the
>     context dynamically to the value of the host in the From: header.
>
>
> Yeah, I can't find any reference anywhere in the code base to 
> "context" ever being equal to "any". I specifically checked 
> mod_dialplan_xml.c and the only place in the code that uses the word 
> "any" is in reference to the "regex" tag. Abaci, please feel free to 
> expunge that information from the wiki. If you can test what Anthony 
> mentioned and confirm that it works then that would be a handy item to 
> add to the wiki.
>
> Thanks,
> MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From acrow at integrafin.co.uk  Tue Apr  3 01:29:36 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 02 Apr 2012 22:29:36 +0100
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
In-Reply-To: <CAKbxfG-x=e2ty2+6UYPpOWDqUdR6eRq7Sb-KDT-c-pcHYhOVdQ@mail.gmail.com>
References: <020701cd1109$037d8cf0$0a78a6d0$@com>
	<4F7A0B04.2070405@integrafin.co.uk>
	<CAKbxfG-x=e2ty2+6UYPpOWDqUdR6eRq7Sb-KDT-c-pcHYhOVdQ@mail.gmail.com>
Message-ID: <4F7A1A40.3050406@integrafin.co.uk>

On 02/04/12 22:02, Anthony Minessale wrote:
> I tracked this down to a race on startup where the core has not loaded
> the ACL lists but sofia tries to use them for NAT detection.
> I'm sure this is not something new but it should be fixed in HEAD.  (I
> WISH SOMEONE WOULD HAVE OPENED A JIRA TO REFERENCE)
>
>
>

Sorry, I did only notice it today :-(

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From cmrienzo at gmail.com  Tue Apr  3 01:42:12 2012
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Mon, 2 Apr 2012 17:42:12 -0400
Subject: [Freeswitch-users] Strange "timer_test" result
In-Reply-To: <CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F15080613@cantor.std.visionutv.se>
	<CAO4LeU08NY0r_Ztsm9SrOe2xajBikcmoH7WUWzo1AgRj0psmPg@mail.gmail.com>
	<CAKbxfG8mTk9v8KfP+ZBpgf8poVq7NV6HTvr-sxGEZbora_VLLw@mail.gmail.com>
	<CAO4LeU18R_MgkKU=UL1t=X2qePyuKRQ9FZTssVTkYyvHnXDRkQ@mail.gmail.com>
	<CAKbxfG9rNPkqhDsB89bMydEayfmskDUhxnu9-n7JoT9ELnWyRA@mail.gmail.com>
	<CAO4LeU3Te+xPZo0BVUDX2k3pN2aW7x-o3XvBvvqtHg8ao+CCjg@mail.gmail.com>
	<CAD-qRkiV4qn3B4NNNdqXrR=mSSpbeNx=MeN8Ty8ChL=REwoCrg@mail.gmail.com>
	<CAO4LeU2a8-417E6aPgb5=_tc1ZZ-Xmai1Fyn=WZ+3pgrYv4cYQ@mail.gmail.com>
	<CAD-qRkhizVe+QsRHcgEseYbPhe2h7=VvtkMj1Nu0vLa00imwSw@mail.gmail.com>
	<CAO4LeU1ppjAo5dTgPywBOeGH0mn9pZC+ov+uu2LNNc6VxyDQcA@mail.gmail.com>
	<0E0EE80C-C059-45CD-BBD4-E0F988C53283@gmail.com>
	<CAO4LeU0F6kcH0tgpEx9039Xf45we2ZFLD_BHU_puhaEyY5mH7Q@mail.gmail.com>
Message-ID: <F615FA10-493C-4AF5-ABE1-DB7E3AE8CA67@gmail.com>

Ok I'll look at it.  



On Apr 2, 2012, at 16:38, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Very satisfied with mod_posix_timer (with this I have solved the issue of soft timer) but the problem of crash is present also with latest version.
> 
> I have open a jira:
> 
> http://jira.freeswitch.org/browse/FS-4064 
> 
> Stephen
> 
> On Tue, Mar 27, 2012 at 10:52 PM, <cmrienzo at gmail.com> wrote:
> No big deal.  I fixed a few issues a while back, so maybe your problem is resolved.  Check the revision history to see if the changes happened after your crash. 
> 
> 
> 
> 
> On Mar 27, 2012, at 14:24, Stephen Wilde <wstephen80 at gmail.com> wrote:
> 
>> The core dump was pretty unusable (stack corrupted) so I had some difficult to report to jira a useful info.
>> 
>> On Tue, Mar 27, 2012 at 7:14 PM, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>> Please report crashes in jira, otherwise they'll never get fixed.
>> 
>> 
>> 
>> On Tue, Mar 27, 2012 at 12:34 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>> I have tried to use mod_posix_timer but with this timer I had some FS crashes (segmentation fault) with core dump so I'm moved back to soft timers.
>> 
>> Stephen
>> 
>> On Tue, Mar 27, 2012 at 6:28 PM, Christopher Rienzo <cmrienzo at gmail.com> wrote:
>> You can also try mod_posix_timer if you have it built.  My CentOS 5.4 server performs better under load using it with heavy-timer vs using the default timer.
>> 
>> "timer_test 20 50 posix"
>> 
>> 
>> 
>> 
>> On Tue, Mar 27, 2012 at 12:07 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
>> Ok, I'll try disabling the nanosleep.
>> 
>> In server.2 the current load of the box is 35% and after FreeSwitch (at 750% of cpu), in top there is Sangoma Transcoding server (4% of cpu).
>> 
>> 
>> The result of "timer_test 20 50" is:
>> 
>> http://pastebin.freeswitch.org/18756
>> 
>> 
>> Here the result of "timer_test 10 50" :
>> 
>> http://pastebin.freeswitch.org/18757
>> 
>> It's normal that only the timer_test on 20 is bad where 10ms is good and also is good 40,60,120?:
>> 
>> Stephen
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From sharad at coraltele.com  Tue Apr  3 11:14:36 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 3 Apr 2012 12:44:36 +0530
Subject: [Freeswitch-users] Session-wise Language Setting
References: <mailman.6169.1332986617.13443.freeswitch-users@lists.freeswitch.org><4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad><4F746EFF.3050305@communicatefreely.net><00C2DD790B254A1BACD50CAA66B123B9@sharad><1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se><12BA6FAF92674DEE9650CF4829B95DF0@sharad>	<CAKzWOxWus_Ryo97ARrJYbFjaJrmPEUxDF169iqzLKbw7AaW+DQ@mail.gmail.com><13D710C546A14C09801E1CC46A394BB4@sharad>
	<4F79CA93.3030205@communicatefreely.net>
Message-ID: <5A93FC0084F946DCB75E8F139E53DD40@sharad>

Hello Mr. Tim

Thanks a ton for your valuable suggestion & time.

I think I found the solution of my query. I will definately try this & let 
you know the feedback.

Thanks a lot once again.

Take care n by.

Sharad


----- Original Message ----- 
From: "Tim St. Pierre" <fs-list at communicatefreely.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, April 02, 2012 9:19 PM
Subject: Re: [Freeswitch-users] Session-wise Language Setting


> Hi Sharad,
>
> When you specify a sound file to play like that, it uses a prefix defined 
> (usually) in vars.xml, and it doesn't expand to the language.  The 
> assumption by the designers I assume was that you would change the prefix 
> to your language.
>
> For multi-language systems, you should always use phrase macros.
>
> On our system, we made several custom phrase macros that are very simple, 
> but will select the correct audio file based on the language.
>
> Any time you would normally specify an audio file, you can instead use 
> phrase:my_macro:data
>
> Then, create a phrase macro for each language that will match your data 
> and play the appropriate audio file.
>
> It could be something as simple as this:
> English:
>
> <macro name="ivr-file">
>      <input pattern="^(\w+)$">
>        <match>
>            <action function="play-file" 
> data="$${media_dir}/en/us/callie/ivr/$1wav"/>
>        </match>
>      </input>
> </macro
>
> And then in your Russian phrase macros:
>
> <macro name="ivr-file">
>      <input pattern="^(\w+)$">
>        <match>
>            <action function="play-file" 
> data="$${media_dir}/ru/RU/elena/ivr/$1wav"/>
>        </match>
>      </input>
> </macro
>
> Be creative here.  Phrase macros are your friend.  I would limit playing 
> audio files to things like sound effects, or other non-verbal recordings.
>
>
> Sharad Garg wrote:
>>
>> Hello Mr. MC
>>
>> Thanks for the link.....seems working.
>>
>> But observed another point i.e. it works only for the speech phrase 
>> macros.
>>
>> When I write a simple xml dialplan like this -
>>
>>     <extension name="lang_test">
>>          <condition field="destination_number" expression="^(3333)$">
>>               <action application="set" data="bypass_media=true"/>
>>               <action application="answer"/>
>>               <action application="sleep" data="1000"/>
>>               <action application="set" data="default_language=ru"/>
>>               <action application="playback" data="ivr/test.wav"/>
>>          </condition>
>>     </extension>
>>
>> This dialplan still plays the test.wav which is in 
>> <base_dir>/sounds/en/us/callie/ivr/8000/test.wav. While it should play 
>> the test.wav from <base_dir>/sounds/ru/RU/elena/ivr/8000/test.wav. & at 
>> the same time, speech phrase macro use the 'ru' language.
>>
>> So when I did my googling, I found that the language settings is only for 
>> speech phrase macros. It is not implemented in Freeswitch to use the 
>> desired languages at all places.
>>
>> So need to know is there any latest development on this ?
>>
>> Thanks in advance.
>>
>> Regards
>>
>> Sharad
>>
>>
>>     ----- Original Message -----
>>     *From:* Michael Collins <mailto:msc at freeswitch.org>
>>     *To:* FreeSWITCH Users Help
>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>     *Sent:* Saturday, March 31, 2012 1:15 AM
>>     *Subject:* Re: [Freeswitch-users] Session-wise Language Setting
>>
>>
>>
>>     On Fri, Mar 30, 2012 at 5:13 AM, Sharad Garg <sharad at coraltele.com
>>     <mailto:sharad at coraltele.com>> wrote:
>>
>>         Thanks Peter...yes it works.
>>
>>         Just a last query - Is there any API for setting the same so
>>         that it can be
>>         used from Javascript.
>>
>>         Best Regards
>>         Sharad
>>
>>
>>     http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar
>>
>> ------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> 



From peter.olsson at visionutveckling.se  Tue Apr  3 12:25:50 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 3 Apr 2012 08:25:50 +0000
Subject: [Freeswitch-users] FS shutdown not closing TCP port
	causes	TIME_WAIT problems
Message-ID: <1FFF97C269757C458224B7C895F35F150870A7@cantor.std.visionutv.se>

By the way - why do you restart FS every night, and do you really need to detect if the process has died and restart it? If you're trying to solve a problem in FS by restarting, it would be much better if a Jira was submitted, so it can be fixed.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mario G
Skickat: den 2 april 2012 22:33
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS shutdown not closing TCP port causes TIME_WAIT problems

That's what I was thinking, trying to figure out how now. Will post a solution when I get one.

On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:

>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
> another MAC specific thing.
> I am not really sure what you can do since you tend to find issues 
> deep inside libsofia-sip and its portability to OSX.
> Maybe make your restart script sleep a bit?
> 
> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
> su_setreuseaddr(socket, 1);
> 
> 
> 
> 
> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
>> Mario G
>> _____________________________________________________________________
>> ____ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>>  
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> ers
>> http://www.freeswitch.org
> 
> 
> 
> --
> Anthony Minessale II
> 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f7a0b6432767969573744!



From anton.jugatsu at gmail.com  Tue Apr  3 13:40:08 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 3 Apr 2012 13:40:08 +0400
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
 TIME_WAIT problems
In-Reply-To: <1FFF97C269757C458224B7C895F35F150870A7@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F150870A7@cantor.std.visionutv.se>
Message-ID: <CANBhyPCiJ+xTXO4wmAu9wWMDG_i65iXNeZWvi-_BU+h9yRKmhw@mail.gmail.com>

Or do not restart at all :)

3 ?????? 2012 ?. 12:25 ???????????? Peter Olsson <
peter.olsson at visionutveckling.se> ???????:

> By the way - why do you restart FS every night, and do you really need to
> detect if the process has died and restart it? If you're trying to solve a
> problem in FS by restarting, it would be much better if a Jira was
> submitted, so it can be fixed.
>
> /Peter
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] F?r Mario G
> Skickat: den 2 april 2012 22:33
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] FS shutdown not closing TCP port causes
> TIME_WAIT problems
>
> That's what I was thinking, trying to figure out how now. Will post a
> solution when I get one.
>
> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>
> >> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
> > another MAC specific thing.
> > I am not really sure what you can do since you tend to find issues
> > deep inside libsofia-sip and its portability to OSX.
> > Maybe make your restart script sleep a bit?
> >
> > libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
> > su_setreuseaddr(socket, 1);
> >
> >
> >
> >
> > On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
> >> If FS is shutdown and started too quickly, the internal profile ip/port
> stays in use and FS will fail to start it. After searching on the wen I
> found many references to TIME_WAIT but all fixes were to the program
> source. So I am wondering if when FS/sofia shuts down they are not
> releasing the ip/port for internal. If it's true, I found some source
> changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff
> like
> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
> Apparently there is no way to free up the port manually. Help/comments
> anyone?
> >> Mario G
> >> _____________________________________________________________________
> >> ____ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> ers
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4f7a0b6432767969573744!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Daniel.Knaggs at realitysolutions.co.uk  Tue Apr  3 14:23:23 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Tue, 3 Apr 2012 10:23:23 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05A3FE@RSV16.realitysolutions.local>

OK, I?ve added the BDA?s to the ISDN dialplan and removed my binding for parking calls (no-one uses that apart from me ? I find it very useful!).

Let?s see what happens?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 02 April 2012 16:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

Maybe bind_digit_action would capture these for you. You could try binding something simple, like a log function:

<action application="bind_digit_action" data="rogue_dtmf,~\w+,exec:log,INFO Rogue digit(s): ${last_matching_digits}"/>
<action application="digit_action_set_realm" data="rogue_dtmf"/>

You'll need to experiment to make sure that regex works. If not you could possibly do each digit explicity, like this:
<action application="bind_digit_action" data="rogue_dtmf,A,exec:log,INFO Rogue digit: DTMF 'A' received"/>
<action application="bind_digit_action" data="rogue_dtmf,B,exec:log,INFO Rogue digit: DTMF 'B' received"/>
<action application="bind_digit_action" data="rogue_dtmf,C,exec:log,INFO Rogue digit: DTMF 'C' received"/>
<action application="bind_digit_action" data="rogue_dtmf,D,exec:log,INFO Rogue digit: DTMF 'D' received"/>

I don't honestly know if this will work, but I'd like to see what happens in your case. FYI, you can read up on bind_digit_action (AKA 'bda') here:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action

-MC
On Mon, Apr 2, 2012 at 5:29 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Interesting, how would I do this?  Can?t use ?bind_meta_app? since that only works with numbers?  The tones appear from the caller not the callee (leg A).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Wesley Akio
Sent: 02 April 2012 12:57

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls


If all you wish is to purge the tones from your stream you can simply capture them in the dialplan, no need to mess with the source...

Best,

Wesley Akio
Em 02/04/2012 05:17, "Daniel Knaggs" <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> escreveu:
I appreciate your honesty, Gabe.  But seriously, is that the reason no-one else is responding or is it because no-one has any ideas?

Unfortunately, I don?t (and can?t) control what appears on my signature as it?s added by our e-mail gateway not my e-mail client (if it can?t see a signature on the e-mail it will add it).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Gabriel Gunderson
Sent: 31 March 2012 09:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Any ideas, anyone?

I'll be honest. I haven't followed the thread because of the loud, in-your-face advertisement in the email footer. Nobody is going to make you remove it, but they might respond to more of your emails if you show them that you respect the list by not cluttering it with noise.

Just my 2 cents.  And of course good luck on the issue.



[cid:image5fa413.PNG at 9c8709cb.4997c030]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:imagefa91bb.PNG at 9afe2a85.4ea36f51]





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From sdame at 207me.com  Tue Apr  3 14:49:43 2012
From: sdame at 207me.com (Stephen Dame)
Date: Tue, 3 Apr 2012 06:49:43 -0400
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
Message-ID: <009001cd1187$7b5c34f0$72149ed0$@com>



I currently accept any "75xxx" numbers  and transfer to a conference with
pin with no problem, however currently if they mistype the number they end
up creating a new conference and sit there all by their self waiting.

 

I want to setup dialpllan entry that only allows a user to join a conference
that already exists and is active.

 

Have read thru the wiki, and the 2  FreeSWITCH books looking for examples on
how to do this without  creating database tables to store conference state.

 

In reading, looks like I can use ${} and get a "conference list" and with
regex patterns might be able to check for existence of valid conference ID
before joining them.

 

Does anyone have a simple approach to do this?   

 

Thanks

Stephen

 

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From gcd at i.ph  Tue Apr  3 15:19:53 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 3 Apr 2012 19:19:53 +0800
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
In-Reply-To: <009001cd1187$7b5c34f0$72149ed0$@com>
References: <009001cd1187$7b5c34f0$72149ed0$@com>
Message-ID: <CAHKqMb6sVwRm4=qwdJ1QBWDy18V1fkKLN=HAXX-ANNaKdPMVyA@mail.gmail.com>

i can suggest:

1) make the PIN resemble the conference number
2) make an announcement of the conference number using "enter-sound"
parameter

http://wiki.freeswitch.org/wiki/Mod_conference

-nandy

On Tue, Apr 3, 2012 at 6:49 PM, Stephen Dame <sdame at 207me.com> wrote:

> I currently accept any ?75xxx? numbers  and transfer to a conference with
> pin with no problem, however currently if they mistype the number they end
> up creating a new conference and sit there all by their self waiting.****
>
> ** **
>
> I want to setup dialpllan entry that only allows a user to join a
> conference that already exists and is active.****
>
> ** **
>
> Have read thru the wiki, and the 2  FreeSWITCH books looking for examples
> on how to do this without  creating database tables to store conference
> state?****
>
> ** **
>
> In reading, looks like I can use ${} and get a ?conference list? and with
> regex patterns might be able to check for existence of valid conference ID
> before joining them.****
>
> ** **
>
> Does anyone have a simple approach to do this?   ****
>
> ** **
>
> Thanks****
>
> Stephen****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Tue Apr  3 15:33:13 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 3 Apr 2012 14:33:13 +0300
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
In-Reply-To: <009001cd1187$7b5c34f0$72149ed0$@com>
References: <009001cd1187$7b5c34f0$72149ed0$@com>
Message-ID: <CALNZuwmgaP09os=_Dm5HvFPdZg85AkwrEfRqvG05CXvqAu1Dxg@mail.gmail.com>

A lua script should be able to handle that easier than doing it in the
dialplan.
But yeah, a regex of an api call to a list sounds like it would work.

Also - how does the FIRST person get in without this same safety mechanism
kicking in?
-Avi


On Tue, Apr 3, 2012 at 1:49 PM, Stephen Dame <sdame at 207me.com> wrote:

> I currently accept any ?75xxx? numbers  and transfer to a conference with
> pin with no problem, however currently if they mistype the number they end
> up creating a new conference and sit there all by their self waiting.****
>
> ** **
>
> I want to setup dialpllan entry that only allows a user to join a
> conference that already exists and is active.****
>
> ** **
>
> Have read thru the wiki, and the 2  FreeSWITCH books looking for examples
> on how to do this without  creating database tables to store conference
> state?****
>
> ** **
>
> In reading, looks like I can use ${} and get a ?conference list? and with
> regex patterns might be able to check for existence of valid conference ID
> before joining them.****
>
> ** **
>
> Does anyone have a simple approach to do this?   ****
>
> ** **
>
> Thanks****
>
> Stephen****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From b2m at a-cti.com  Tue Apr  3 17:36:14 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Tue, 3 Apr 2012 19:06:14 +0530
Subject: [Freeswitch-users] RTMP - call delay
Message-ID: <CAP2fbbce6Lrimy0+pibEJBxOMtfc4NBuJ5t0Ld-o_toe2rB=hw@mail.gmail.com>

Hi all,

Is there a way to reduce call delay for rtmp? trying to dial from flex
module. Getting minimum 10 sec delay.

Thanks for your help!!

--Bala
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From anthony.minessale at gmail.com  Tue Apr  3 19:06:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 3 Apr 2012 10:06:24 -0500
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
In-Reply-To: <CALNZuwmgaP09os=_Dm5HvFPdZg85AkwrEfRqvG05CXvqAu1Dxg@mail.gmail.com>
References: <009001cd1187$7b5c34f0$72149ed0$@com>
	<CALNZuwmgaP09os=_Dm5HvFPdZg85AkwrEfRqvG05CXvqAu1Dxg@mail.gmail.com>
Message-ID: <CAKbxfG8Efv71Q13nDbwheCV8CpawX7jmXC6Q3zdx_QGe-KPxvw@mail.gmail.com>

Try this:

<condition field="${conference ${destination_number} list count}"
expression="^\d+">




On Tue, Apr 3, 2012 at 6:33 AM, Avi Marcus <avi at avimarcus.net> wrote:

> A lua script should be able to handle that easier than doing it in the
> dialplan.
> But yeah, a regex of an api call to a list sounds like it would work.
>
> Also - how does the FIRST person get in without this same safety mechanism
> kicking in?
> -Avi
>
>
> On Tue, Apr 3, 2012 at 1:49 PM, Stephen Dame <sdame at 207me.com> wrote:
>
>> I currently accept any ?75xxx? numbers  and transfer to a conference with
>> pin with no problem, however currently if they mistype the number they end
>> up creating a new conference and sit there all by their self waiting.****
>>
>> ** **
>>
>> I want to setup dialpllan entry that only allows a user to join a
>> conference that already exists and is active.****
>>
>> ** **
>>
>> Have read thru the wiki, and the 2  FreeSWITCH books looking for examples
>> on how to do this without  creating database tables to store conference
>> state?****
>>
>> ** **
>>
>> In reading, looks like I can use ${} and get a ?conference list? and with
>> regex patterns might be able to check for existence of valid conference ID
>> before joining them.****
>>
>> ** **
>>
>> Does anyone have a simple approach to do this?   ****
>>
>> ** **
>>
>> Thanks****
>>
>> Stephen****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From Daniel.Knaggs at realitysolutions.co.uk  Tue Apr  3 19:17:57 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Tue, 3 Apr 2012 15:17:57 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com> 
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>

Update, it would appear (so far) that this has worked.

I can see in the log numerous ?Rogue digit: DTMF 'A' received? entries (and also for ?C?) and I?ve asked the users if they have heard any more tones and they haven?t.


Thanks, Michael.   So we?ve managed to work around it, but how can I track down what is generating these tones to begin with?  Is my earlier spectrum analysis of a call any good?


[cid:image83f6e1.PNG at df16e8b7.49bddf57]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:image6ddd5c.PNG at d6dd7aa3.468d7b5a]


From: Daniel Knaggs
Sent: 03 April 2012 11:23
To: 'FreeSWITCH Users Help'
Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

OK, I?ve added the BDA?s to the ISDN dialplan and removed my binding for parking calls (no-one uses that apart from me ? I find it very useful!).

Let?s see what happens?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Michael Collins
Sent: 02 April 2012 16:44
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

Maybe bind_digit_action would capture these for you. You could try binding something simple, like a log function:

<action application="bind_digit_action" data="rogue_dtmf,~\w+,exec:log,INFO Rogue digit(s): ${last_matching_digits}"/>
<action application="digit_action_set_realm" data="rogue_dtmf"/>

You'll need to experiment to make sure that regex works. If not you could possibly do each digit explicity, like this:
<action application="bind_digit_action" data="rogue_dtmf,A,exec:log,INFO Rogue digit: DTMF 'A' received"/>
<action application="bind_digit_action" data="rogue_dtmf,B,exec:log,INFO Rogue digit: DTMF 'B' received"/>
<action application="bind_digit_action" data="rogue_dtmf,C,exec:log,INFO Rogue digit: DTMF 'C' received"/>
<action application="bind_digit_action" data="rogue_dtmf,D,exec:log,INFO Rogue digit: DTMF 'D' received"/>

I don't honestly know if this will work, but I'd like to see what happens in your case. FYI, you can read up on bind_digit_action (AKA 'bda') here:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action

-MC
On Mon, Apr 2, 2012 at 5:29 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Interesting, how would I do this?  Can?t use ?bind_meta_app? since that only works with numbers?  The tones appear from the caller not the callee (leg A).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Wesley Akio
Sent: 02 April 2012 12:57

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls


If all you wish is to purge the tones from your stream you can simply capture them in the dialplan, no need to mess with the source...

Best,

Wesley Akio
Em 02/04/2012 05:17, "Daniel Knaggs" <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> escreveu:
I appreciate your honesty, Gabe.  But seriously, is that the reason no-one else is responding or is it because no-one has any ideas?

Unfortunately, I don?t (and can?t) control what appears on my signature as it?s added by our e-mail gateway not my e-mail client (if it can?t see a signature on the e-mail it will add it).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Gabriel Gunderson
Sent: 31 March 2012 09:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

On Fri, Mar 30, 2012 at 2:16 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Any ideas, anyone?

I'll be honest. I haven't followed the thread because of the loud, in-your-face advertisement in the email footer. Nobody is going to make you remove it, but they might respond to more of your emails if you show them that you respect the list by not cluttering it with noise.

Just my 2 cents.  And of course good luck on the issue.





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From msc at freeswitch.org  Tue Apr  3 19:22:43 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 3 Apr 2012 08:22:43 -0700
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
In-Reply-To: <CAKbxfG8Efv71Q13nDbwheCV8CpawX7jmXC6Q3zdx_QGe-KPxvw@mail.gmail.com>
References: <009001cd1187$7b5c34f0$72149ed0$@com>
	<CALNZuwmgaP09os=_Dm5HvFPdZg85AkwrEfRqvG05CXvqAu1Dxg@mail.gmail.com>
	<CAKbxfG8Efv71Q13nDbwheCV8CpawX7jmXC6Q3zdx_QGe-KPxvw@mail.gmail.com>
Message-ID: <CAKzWOxUp3o6ToB2bRTr7W6O6NqeE8c-Db+XYDh55MzPUqNW0WA@mail.gmail.com>

On Tue, Apr 3, 2012 at 8:06 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Try this:
>
> <condition field="${conference ${destination_number} list count}"
> expression="^\d+">
>
>
Stephen, as you can see you were on the right track w/ doing an API call to
see if the conference exists. Just one quick note: be sure to put the full
name of your conference in the above command. For example, in the default
dialplan the conference room names have "-${domain}" appended to them. If
you do anything like that when you create your conference then be sure to
account for it. Here's an example of a few conferences on a vanilla install
of FreeSWITCH:

freeswitch at default> conference list
Conference 3300-192.168.1.79 (1 member rate: 48000)
2;loopback/9664-a;9f09f968-7d9e-11e1-b22d-0bde109daa55;Outbound
Call;9664;hear|speak|floor;0;0;0;300
Conference 3100-192.168.1.79 (1 member rate: 16000)
1;loopback/9664-a;91e680da-7d9e-11e1-b224-0bde109daa55;Outbound
Call;9664;hear|speak|floor;0;0;0;300

Consider "3100" - the person dials 3100 to get into the conference but the
actual name of the conference is "3100-192.168.1.79" where that IP address
is my ${domain} value. In that case you could do something like this:

<condition field="${conference ${destination_number}-${domain} list count}"
expression="^\d+">

If you run into any trouble then do the commands at fs_cli and see what
happens. Here's some examples:

freeswitch at default> conference 3100 list count
Conference 3100 not found

freeswitch at default> conference 3100-192.168.1.79 list count
1
freeswitch at default> conference 3100-192.168.1.79 list
1;loopback/9664-a;91e680da-7d9e-11e1-b224-0bde109daa55;Outbound
Call;9664;hear|speak|talking|floor;0;0;0;300


In any case, we invite you to experiment, get it working, and then update
the wiki. Oh, and thank you for buying the two FS books!

-MC
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From msc at freeswitch.org  Tue Apr  3 19:30:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 3 Apr 2012 08:30:32 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>

On Tue, Apr 3, 2012 at 8:17 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

>   Update, it would appear (so far) that this has worked.****
>
> ** **
>
> I can see in the log numerous ?Rogue digit: DTMF 'A' received? entries
> (and also for ?C?) and I?ve asked the users if they have heard any more
> tones and they haven?t.****
>
> ** **
>
> ** **
>
> Thanks, Michael.   So we?ve managed to work around it, but how can I track
> down what is generating these tones to begin with?  Is my earlier spectrum
> analysis of a call any good?
>

I don't think we ever truly answered the question of whether or not these
rogue digits are phantoms. Remind me: how did you capture the audio on
which you did the spectrum analysis? I recommend using ftdm dump, which is
nice because it captures raw audio in either direction, and if I understand
correctly it is what's "on the line" - if those rogue digits are coming
from the other end they'll show up in the raw dump. If those digits are not
in the raw audio then most likely you're looking at a false positive from
the DTMF detector, in which case you've got a bit of a conundrum because
you've got hardware from Digium and software from Sangoma and you need to
figure out where to begin looking...

BTW, feel free to keep the BDA's in there and just add your transfer digits
to the digit action realm and you can avoid using bind meta app. BDA is the
preferred method nowadays.

-MC
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From mario_fs at mgtech.com  Tue Apr  3 19:53:22 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 08:53:22 -0700
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
	TIME_WAIT problems
In-Reply-To: <E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
Message-ID: <C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>

For anyone who may run into this see "FS shutdown not closing TCP port causes TIME_WAIT problems" for follow up and solution.
On Apr 2, 2012, at 1:33 PM, Mario G wrote:

> That's what I was thinking, trying to figure out how now. Will post a solution when I get one.
> 
> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
> 
>>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>> another MAC specific thing.
>> I am not really sure what you can do since you tend to find issues
>> deep inside libsofia-sip and its portability to OSX.
>> Maybe make your restart script sleep a bit?
>> 
>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>> su_setreuseaddr(socket, 1);
>> 
>> 
>> 
>> 
>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
>>> Mario G
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Tue Apr  3 19:55:52 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 08:55:52 -0700
Subject: [Freeswitch-users] Internal profile not started at FS
	startup	if extension uses TCP
In-Reply-To: <8454EDB4-1A55-4D10-9B04-1E886E976B23@mgtech.com>
References: <101A267D-2CF3-4187-A1A9-EC234F484EBC@mgtech.com>
	<CAKzWOxVPc=YemMiORPk=9-+Jkq=HcUyF18MHAMEmDEbyVpg+gw@mail.gmail.com>,
	<EABD7E78-A859-4D02-8D99-62824FBFDB81@mgtech.com>
	<0AB00DEE-88D8-44E8-9F34-00A1218565BC@visionutveckling.se>
	<8103D182-9384-4861-9A1E-27D03E53EF68@mgtech.com>
	<8454EDB4-1A55-4D10-9B04-1E886E976B23@mgtech.com>
Message-ID: <4E9F6212-C2D5-424A-9DFF-33337BE95923@mgtech.com>

For anyone who may run into this see "FS shutdown not closing TCP port causes TIME_WAIT problems" for follow up and solution.

On Apr 2, 2012, at 12:42 PM, Mario G wrote:

> Looks like I know what it is but have no idea how to fix it. Multiple tests are reproducible. I did a NETSTAT -p TCP to watch the .7 address (oddly port 5060 show up as "sip" in the netstat command on OSX). When FS comes down this port stays open for about 30 seconds which is longer than the FS restart takes so sofia says .7:5060 is in use. I also tied setting keep alive to 0. I now finally figured out the problem but.... Does anyone have an idea how to force the ip/port closed sooner? 
> 
> On Apr 2, 2012, at 11:59 AM, Mario G wrote:
> 
>> There is a plist that starts FS when it is down, then another plist shuts it down once a day, the startup plist causes it to restart. It's exactly the ones I put in the FS OSX install wiki (speaking of I will update this week). It's odd that it ONLY happens when an extension is set to TCP. Also, I am assuming this is the problem... I will have to retest without the TCP extensions to see it it is.
>> 
>> On Apr 2, 2012, at 11:47 AM, Peter Olsson wrote:
>> 
>>> It means port 5060 is busy (probably by the old FS instance). How do you restart FS?
>>> 
>>> /Peter
>>> 
>>> 2 apr 2012 kl. 20:34 skrev "Mario G" <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>>:
>>> 
>>> It is a plist on OSX. Has been fine since 2010. I duplicated the issue by setting the in the plist, here is a replacement since I turned on all kinds of logging and finally found a message. Here is an update:
>>> 
>>> A TCP extension causes already in use MSG and  Internal profile fails
>>> 
>>> I added an iPad softphone (tested SoftPhone and Bria) that requires TCP and all works fine except: When FS is restarted 2am each day the INTERNAL profile is not activated causing all internal extensions to stay offline. If  the TCP extension is off or set UDP all works.  Anytime after the restart I can "sofia profile internal start" to fix it. I am on Saturdays (03-30) GIT (to test the new 1.2). The only thing I can find is the "Address already in use" below which is shortly followed by the error creating internal. The .7 address is FS and there is nothing else using it. Any ideas why it says in use ONLY when an extension is set for TCP?
>>> Mario G
>>> 
>>> tport_bind_server(0x1003a9850): calling tport_listen for tcp
>>> nta: master transport created
>>> tport_alloc_primary(0x1003a9850): new primary tport 0x1003993e0
>>> tport_listen(0x1003a9850): bind(pf=2 tcp/[100.234.1.7]:5060): Address already in use   <---------------
>>> nta: bind(100.234.1.7:5060;transport=*): Address already in use
>>> nua: initializing SIP stack failed
>>> nua: nua_stack_deinit: entering
>>> tport_bind_server(0x1006dd850) to */100.234.1.7:5078/sip
>>> sres_sofia_update(0x1003a7e10, -1, -1)
>>> sres_sofia_update(0x0, -1, -1)
>>> tport_destroy(0x1003a9850)
>>> tport_bind_server(0x1006dd850): calling tport_listen for udp
>>> tport_alloc_primary(0x1006dd850): new primary tport 0x1003a9ba0
>>> su_kqueue_port_deinit(0x10039e350) called
>>> tport_listen(0x1003a9ba0): listening at udp/100.234.1.7:5078/sip
>>> tport_bind_server(0x1006dd850): calling tport_listen for tcp
>>> tport_alloc_primary(0x1006dd850): new primary tport 0x10039e440
>>> tport_listen(0x10039e440): listening at tcp/100.234.1.7:5078/sip
>>> nta: bound to (100.234.1.7:5078;transport=*)
>>> nta: agent_init_via: SIP/2.0/udp 100.234.1.7:5078 (sip)
>>> nta: agent_init_via: SIP/2.0/tcp 100.234.1.7:5078 (sip)
>>> nta: Via fields initialized
>>> nta: Contact header created
>>> 2012-04-02 11:00:35.147184 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal   <---------------------
>>> nua_register: Adding contact URL '100.234.1.7' to list.
>>> tport_wakeup_pri(0x1003a9ba0): events IN
>>> tport_recv_event(0x1003a9ba0)
>>> tport_recv_iovec(0x1003a9ba0) msg 0x1003a74e0 from (udp/100.234.1.7:5078) has 20 bytes, veclen = 1
>>> 
>>> 
>>> 
>>> 
>>> On Apr 2, 2012, at 11:08 AM, Michael Collins wrote:
>>> 
>>> What is the 2am restart? Is that a cron job or...?
>>> 
>>> On Mon, Apr 2, 2012 at 9:14 AM, Mario G <mario_fs at mgtech.com<mailto:mario_fs at mgtech.com>> wrote:
>>> I added an iPad softphone that requires TCP to converse battery. Everything works except: FS is restarted 2am each day but does not activate the INTERNAL profile, this causes all internal extensions to stay offline. If  the TCP extension is off all works. Odd: I cannot reproduce this at any other time manually, it only occurs at the 2am restart! So debugging has been a bugger. Anytime after 2am I can "sofia profile internal start" and all is well. I ran with sofia loglevel all 9 and the TCP devices IPs did not show up in the log at all and NO MENTION of INTERNAL in the log. It's as if the file is missing, there is nothing in the FS log to provide a hint. I searched the FS wiki and web and found a TCP bug mentioned march 2011 but I updated FS to Saturdays (03-30) GIT (to test the new 1.2 anyway) and no change. BTW, I tried 2 different clients (SoftPhone and Bria) and both had the problem. Anyone have an idea what to try next?
>>> Mario G
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> !DSPAM:4f79ee9c32766092896665!
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> !DSPAM:4f79ee9c32766092896665!
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Tue Apr  3 19:57:12 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 08:57:12 -0700
Subject: [Freeswitch-users] FS shutdown not closing TCP port causes
	TIME_WAIT problems
In-Reply-To: <CANBhyPCiJ+xTXO4wmAu9wWMDG_i65iXNeZWvi-_BU+h9yRKmhw@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F150870A7@cantor.std.visionutv.se>
	<CANBhyPCiJ+xTXO4wmAu9wWMDG_i65iXNeZWvi-_BU+h9yRKmhw@mail.gmail.com>
Message-ID: <3F7E17BF-4965-4378-8583-ED48A4AFD02D@mgtech.com>

Need to restart due to memory leak bug that's been on Jira for years. May be gone now need to test more.

On Apr 3, 2012, at 2:40 AM, Anton Kvashenkin wrote:

> Or do not restart at all :)
> 
> 3 ?????? 2012 ?. 12:25 ???????????? Peter Olsson <peter.olsson at visionutveckling.se> ???????:
> By the way - why do you restart FS every night, and do you really need to detect if the process has died and restart it? If you're trying to solve a problem in FS by restarting, it would be much better if a Jira was submitted, so it can be fixed.
> 
> /Peter
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mario G
> Skickat: den 2 april 2012 22:33
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] FS shutdown not closing TCP port causes TIME_WAIT problems
> 
> That's what I was thinking, trying to figure out how now. Will post a solution when I get one.
> 
> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
> 
> >> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
> > another MAC specific thing.
> > I am not really sure what you can do since you tend to find issues
> > deep inside libsofia-sip and its portability to OSX.
> > Maybe make your restart script sleep a bit?
> >
> > libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
> > su_setreuseaddr(socket, 1);
> >
> >
> >
> >
> > On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
> >> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
> >> Mario G
> >> _____________________________________________________________________
> >> ____ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> ers
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4f7a0b6432767969573744!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Tue Apr  3 20:17:45 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 09:17:45 -0700
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
	causes TIME_WAIT problems
In-Reply-To: <C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
Message-ID: <350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>

I am posting this in case someone else runs into this, especially since I found it may not be Mac OSX specific (I was able to duplicate on SUSE although not as much). 

Both of the following conditions must exist:
1. You have at least one TCP local extension. In this case, due to Bria, etc. on an iPad, using TCP reduced battery usage since no need to run in background.
2. FreeSwitch is recycled very quickly, in this case down/up within 30 seconds because it was automated at 2am each day. I needed it due to a long running sofia memory leak on OSX which may be resolved but I need to test.

Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060 port active (but NOT open) on the TCP stack, in my case it was 30 seconds but I found the time could be different on other platforms. When FreeSwitch comes back up the "internal.xml" profile fails because port 5060 is in use. If FS is brought up AFTER the TIME_WAIT goes away there is no problem, but in my case it was very quick each night to minimize the outage.

Solution: I changed the OSX plist that auto restarts FS to run a script that starts FS, this makes no difference except a script is more flexible. Rather than use an arbitrary sleep time I check for port 5060 which is more reliable and faster. I tested this many times using different scenarios and all worked. Note I tried to use the "lsof -i TCP:5060" command but it shows only OPEN ports so it won't work. 

#!/bin/sh
while : ; do
   a=$(/usr/sbin/netstat -p tcp)
   if [[ "$a" != *sip* ]] ; then
     break
   fi
   echo "Port 5060 still in use - waiting for it to free up"
   echo "${a}"
   /bin/sleep 3
done
/usr/local/freeswitch/bin/freeswitch -nc -nonat

#   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
#   if [ "$a" = "" ] ; then



On Apr 3, 2012, at 8:53 AM, Mario G wrote:

> For anyone who may run into this see "FS shutdown not closing TCP port causes TIME_WAIT problems" for follow up and solution.
> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
> 
>> That's what I was thinking, trying to figure out how now. Will post a solution when I get one.
>> 
>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>> 
>>>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>>> another MAC specific thing.
>>> I am not really sure what you can do since you tend to find issues
>>> deep inside libsofia-sip and its portability to OSX.
>>> Maybe make your restart script sleep a bit?
>>> 
>>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>>> su_setreuseaddr(socket, 1);
>>> 
>>> 
>>> 
>>> 
>>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> If FS is shutdown and started too quickly, the internal profile ip/port stays in use and FS will fail to start it. After searching on the wen I found many references to TIME_WAIT but all fixes were to the program source. So I am wondering if when FS/sofia shuts down they are not releasing the ip/port for internal. If it's true, I found some source changes to solve it. Google "TIME-WAIT" and you'll find lots of good stuff like http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait. Apparently there is no way to free up the port manually. Help/comments anyone?
>>>> Mario G
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> -- 
>>> Anthony Minessale II
>>> 
>> 

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From sdame at 207me.com  Tue Apr  3 20:49:05 2012
From: sdame at 207me.com (Stephen Dame)
Date: Tue, 3 Apr 2012 12:49:05 -0400
Subject: [Freeswitch-users] enter only existing conferences in dialplan.
In-Reply-To: <CAKbxfG8Efv71Q13nDbwheCV8CpawX7jmXC6Q3zdx_QGe-KPxvw@mail.gmail.com>
References: <009001cd1187$7b5c34f0$72149ed0$@com><CALNZuwmgaP09os=_Dm5HvFPdZg85AkwrEfRqvG05CXvqAu1Dxg@mail.gmail.com>
	<CAKbxfG8Efv71Q13nDbwheCV8CpawX7jmXC6Q3zdx_QGe-KPxvw@mail.gmail.com>
Message-ID: <009501cd11b9$af2c7240$0d8556c0$@com>

Thanks, for the help everyone,  I'm off and running with all your advice,
in my case I have an app {bigbluebutton] that creates the initial conf
number using another dial plan entry.

 

This is allowing an external DID to call in, and only join the valid
conference that is already running with no pin needed.  The application
displays the conf number, so users just have to dial, and enter the valid
conf number.

 

Both books are great, and every time I dig into them a learn a bunch more.

 

Regards,

Stephen

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, April 03, 2012 11:06 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] enter only existing conferences in dialplan.

 

Try this:

 

<condition field="${conference ${destination_number} list count}"
expression="^\d+">

 

 

 

On Tue, Apr 3, 2012 at 6:33 AM, Avi Marcus <avi at avimarcus.net> wrote:

A lua script should be able to handle that easier than doing it in the
dialplan.

But yeah, a regex of an api call to a list sounds like it would work.

 

Also - how does the FIRST person get in without this same safety mechanism
kicking in?


-Avi

 

On Tue, Apr 3, 2012 at 1:49 PM, Stephen Dame <sdame at 207me.com> wrote:



I currently accept any "75xxx" numbers  and transfer to a conference with
pin with no problem, however currently if they mistype the number they end
up creating a new conference and sit there all by their self waiting.

 

I want to setup dialpllan entry that only allows a user to join a conference
that already exists and is active.

 

Have read thru the wiki, and the 2  FreeSWITCH books looking for examples on
how to do this without  creating database tables to store conference state.

 

In reading, looks like I can use ${} and get a "conference list" and with
regex patterns might be able to check for existence of valid conference ID
before joining them.

 

Does anyone have a simple approach to do this?   

 

Thanks

Stephen

 

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





 

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900

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From Daniel.Knaggs at realitysolutions.co.uk  Tue Apr  3 19:52:21 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Tue, 3 Apr 2012 15:52:21 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>

Audio was captured using ?record_session? where it didn?t appear in the audio by ?ear?.

?ftdm dump? displays details about the span I tell it to: -

span_id: 1
chan_id: 1
physical_span_id: 1
physical_chan_id: 1
physical_status: ok
physical_status_red: 0
physical_status_yellow: 0
physical_status_rai: 0
physical_status_blue: 0
physical_status_ais: 0
physical_status_general: 0
signaling_status: UP
type: B
state: DOWN
last_state: HANGUP_COMPLETE
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)

etc?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 03 April 2012 16:31
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls


On Tue, Apr 3, 2012 at 8:17 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Update, it would appear (so far) that this has worked.

I can see in the log numerous ?Rogue digit: DTMF 'A' received? entries (and also for ?C?) and I?ve asked the users if they have heard any more tones and they haven?t.


Thanks, Michael.   So we?ve managed to work around it, but how can I track down what is generating these tones to begin with?  Is my earlier spectrum analysis of a call any good?

I don't think we ever truly answered the question of whether or not these rogue digits are phantoms. Remind me: how did you capture the audio on which you did the spectrum analysis? I recommend using ftdm dump, which is nice because it captures raw audio in either direction, and if I understand correctly it is what's "on the line" - if those rogue digits are coming from the other end they'll show up in the raw dump. If those digits are not in the raw audio then most likely you're looking at a false positive from the DTMF detector, in which case you've got a bit of a conundrum because you've got hardware from Digium and software from Sangoma and you need to figure out where to begin looking...

BTW, feel free to keep the BDA's in there and just add your transfer digits to the digit action realm and you can avoid using bind meta app. BDA is the preferred method nowadays.

-MC

[cid:image23c8a0.PNG at a0d20f22.44823835]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:imagee8ba2d.PNG at 2a4e8ae6.4a92bc17]





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From msc at freeswitch.org  Tue Apr  3 21:23:32 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 3 Apr 2012 10:23:32 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>

Oops, mea culpa! I meant "ftdm trace" which will record the audio on a
whole span or a single channel depending on the args.
-MC

On Tue, Apr 3, 2012 at 8:52 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

>   Audio was captured using ?record_session? where it didn?t appear in the
> audio by ?ear?.****
>
> ** **
>
> ?ftdm dump? displays details about the span I tell it to: -****
>
> ** **
>
> span_id: 1****
>
> chan_id: 1****
>
> physical_span_id: 1****
>
> physical_chan_id: 1****
>
> physical_status: ok****
>
> physical_status_red: 0****
>
> physical_status_yellow: 0****
>
> physical_status_rai: 0****
>
> physical_status_blue: 0****
>
> physical_status_ais: 0****
>
> physical_status_general: 0****
>
> signaling_status: UP****
>
> type: B****
>
> state: DOWN****
>
> last_state: HANGUP_COMPLETE****
>
> txgain: 0.00****
>
> rxgain: 0.00****
>
> cid_date:****
>
> cid_name:****
>
> cid_num:****
>
> ani:****
>
> aniII:****
>
> dnis:****
>
> rdnis:****
>
> cause: NONE****
>
> session: (none)****
>
> ** **
>
> etc?****
>
>
>
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From anthony.minessale at gmail.com  Tue Apr  3 22:24:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 3 Apr 2012 13:24:11 -0500
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
 causes TIME_WAIT problems
In-Reply-To: <350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
Message-ID: <CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>

How exactly does it "stop" freeswitch?

On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
> I am posting this in case someone else runs into this, especially since I
> found it may not be Mac OSX specific (I was able to duplicate on SUSE
> although not as much).
>
> Both of the following conditions must exist:
> 1. You have at least one TCP local extension. In this case, due to Bria,
> etc. on an iPad, using TCP reduced battery usage since no need to run in
> background.
> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
> seconds because it was automated at 2am each day. I needed it due to a long
> running sofia memory leak on OSX which may be resolved but I need to test.
>
> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060 port
> active (but NOT open) on the TCP stack, in my case it was 30 seconds but I
> found the time could be different on other platforms. When FreeSwitch comes
> back up the "internal.xml" profile fails because port 5060 is in use. If FS
> is brought up AFTER the TIME_WAIT goes away there is no problem, but in my
> case it was very quick each night to minimize the outage.
>
> Solution: I changed the OSX plist that auto restarts FS to run a script that
> starts FS, this makes no difference except a script is more flexible. Rather
> than use an arbitrary sleep time I check for port 5060 which is more
> reliable and faster. I tested this many times using different scenarios and
> all worked. Note I tried to use the "lsof -i TCP:5060" command but it shows
> only OPEN ports so it won't work.
>
> #!/bin/sh
> while : ; do
> ?? a=$(/usr/sbin/netstat -p tcp)
> ?? if [[ "$a" != *sip* ]] ; then
> ?? ? break
> ?? fi
> ?? echo "Port 5060 still in use - waiting for it to free up"
> ?? echo "${a}"
> ?? /bin/sleep 3
> done
> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>
> # ? a=$(/usr/sbin/lsof -i TCP:5060)? <- only shows open ports
> # ? if [ "$a" = "" ] ; then
>
>
>
> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>
> For anyone who may run into this see "FS shutdown not closing TCP port
> causes TIME_WAIT problems" for follow up and solution.
> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>
> That's what I was thinking, trying to figure out how now. Will post a
> solution when I get one.
>
> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>
> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>
> another MAC specific thing.
>
> I am not really sure what you can do since you tend to find issues
>
> deep inside libsofia-sip and its portability to OSX.
>
> Maybe make your restart script sleep a bit?
>
>
> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>
> su_setreuseaddr(socket, 1);
>
>
>
>
>
> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>
> If FS is shutdown and started too quickly, the internal profile ip/port
> stays in use and FS will fail to start it. After searching on the wen I
> found many references to TIME_WAIT but all fixes were to the program source.
> So I am wondering if when FS/sofia shuts down they are not releasing the
> ip/port for internal. If it's true, I found some source changes to solve it.
> Google "TIME-WAIT" and you'll find lots of good stuff like
> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
> Apparently there is no way to free up the port manually. Help/comments
> anyone?
>
> Mario G
>
> _________________________________________________________________________
>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
>
>
> 
>
> 
>
>
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>
> http://wiki.freeswitch.org
>
> http://www.cluecon.com
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>
> --
>
> Anthony Minessale II
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From admharris at gmx.com  Tue Apr  3 22:43:36 2012
From: admharris at gmx.com (adam harris)
Date: Tue, 03 Apr 2012 20:43:36 +0200
Subject: [Freeswitch-users] collect dtmf events for a timed period
Message-ID: <20120403184336.27640@gmx.com>

How can I set to recevice dtmf events for a timed period in lua. I want to evaluate each key press so this requires consuming events or callback I would guess. Any sugesstions or examples




From joe.jflemmings at gmail.com  Tue Apr  3 22:53:06 2012
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Tue, 3 Apr 2012 11:53:06 -0700
Subject: [Freeswitch-users] Question on runtime.profile_time in
	switch_core_session.c
Message-ID: <CALH5ZramhO1zVUzs3Dmi3j06y0J3VPKG-CSJqjGyt2cJT7JAuw@mail.gmail.com>

Once I restart FreeSwitch I?m able to do 100cps without a problem but once
I go beyond this I get ?503 Maximum Calls In Progress? which is okay due to
my hardware limitation.
My problem is that once I get this SIP message, and restart my tests again
without starting FreeSwitch, I only able to do about 50 and start getting
the  ?503 Maximum Calls In Progress?. I have to restart FreeSWITCH to go
back to 100cps.
After digging in a little I see that in switch_core_session.c ,
runtime.profile_time
has a lower value when there is one call before the restart than after the
restart.

This causes the calls to reach CPU limits faster than when I restart
Freeswitch.





        if (runtime.min_idle_time > 0 && runtime.profile_time <
runtime.min_idle_time) {

                return NULL;

        }



What could be causing this?

Joe
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From mario_fs at mgtech.com  Wed Apr  4 00:47:40 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 13:47:40 -0700
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
	causes TIME_WAIT problems
In-Reply-To: <CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
Message-ID: <0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>

If you mean the effect, it did not stop FS although the results were the same, it prevented the fast FS shutdown/startup from activating the internal profile so none of the phone extensions work. In my case that meant all phones were out from 2am to whenever I activated internal. Not good in case of an emergency. Hope it may be useful to someone, it took me 15 days to find and resolve.

If you mean how is FreeSwitch shutdown, the 2am script runs a script that runs one command: /usr/local/freeswitch/bin/fs_cli -x "shutdown"

On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:

> How exactly does it "stop" freeswitch?
> 
> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>> I am posting this in case someone else runs into this, especially since I
>> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>> although not as much).
>> 
>> Both of the following conditions must exist:
>> 1. You have at least one TCP local extension. In this case, due to Bria,
>> etc. on an iPad, using TCP reduced battery usage since no need to run in
>> background.
>> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>> seconds because it was automated at 2am each day. I needed it due to a long
>> running sofia memory leak on OSX which may be resolved but I need to test.
>> 
>> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060 port
>> active (but NOT open) on the TCP stack, in my case it was 30 seconds but I
>> found the time could be different on other platforms. When FreeSwitch comes
>> back up the "internal.xml" profile fails because port 5060 is in use. If FS
>> is brought up AFTER the TIME_WAIT goes away there is no problem, but in my
>> case it was very quick each night to minimize the outage.
>> 
>> Solution: I changed the OSX plist that auto restarts FS to run a script that
>> starts FS, this makes no difference except a script is more flexible. Rather
>> than use an arbitrary sleep time I check for port 5060 which is more
>> reliable and faster. I tested this many times using different scenarios and
>> all worked. Note I tried to use the "lsof -i TCP:5060" command but it shows
>> only OPEN ports so it won't work.
>> 
>> #!/bin/sh
>> while : ; do
>>    a=$(/usr/sbin/netstat -p tcp)
>>    if [[ "$a" != *sip* ]] ; then
>>      break
>>    fi
>>    echo "Port 5060 still in use - waiting for it to free up"
>>    echo "${a}"
>>    /bin/sleep 3
>> done
>> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>> 
>> #   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
>> #   if [ "$a" = "" ] ; then
>> 
>> 
>> 
>> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>> 
>> For anyone who may run into this see "FS shutdown not closing TCP port
>> causes TIME_WAIT problems" for follow up and solution.
>> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>> 
>> That's what I was thinking, trying to figure out how now. Will post a
>> solution when I get one.
>> 
>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>> 
>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>> 
>> another MAC specific thing.
>> 
>> I am not really sure what you can do since you tend to find issues
>> 
>> deep inside libsofia-sip and its portability to OSX.
>> 
>> Maybe make your restart script sleep a bit?
>> 
>> 
>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>> 
>> su_setreuseaddr(socket, 1);
>> 
>> 
>> 
>> 
>> 
>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>> 
>> If FS is shutdown and started too quickly, the internal profile ip/port
>> stays in use and FS will fail to start it. After searching on the wen I
>> found many references to TIME_WAIT but all fixes were to the program source.
>> So I am wondering if when FS/sofia shuts down they are not releasing the
>> ip/port for internal. If it's true, I found some source changes to solve it.
>> Google "TIME-WAIT" and you'll find lots of good stuff like
>> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
>> Apparently there is no way to free up the port manually. Help/comments
>> anyone?
>> 
>> Mario G
>> 
>> _________________________________________________________________________
>> 
>> 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From bdfoster at endigotech.com  Wed Apr  4 01:09:28 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 3 Apr 2012 17:09:28 -0400
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
 causes TIME_WAIT problems
In-Reply-To: <0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
	<0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
Message-ID: <CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>

fs_cli -x "fsctl shutdown gracefully"

-BDF
On Apr 3, 2012 4:48 PM, "Mario G" <mario_fs at mgtech.com> wrote:

> If you mean the effect, it did not stop FS although the results were the
> same, it prevented the fast FS shutdown/startup from activating the
> internal profile so *none* of the phone extensions work. In my case that
> meant all phones were out from 2am to whenever I activated internal. Not
> good in case of an emergency. Hope it may be useful to someone, it took me
> 15 days to find and resolve.
>
> If you mean how is FreeSwitch shutdown, the 2am script runs a script that
> runs one command: /usr/local/freeswitch/bin/fs_cli -x "shutdown"
>
> On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:
>
> How exactly does it "stop" freeswitch?
>
> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>
> I am posting this in case someone else runs into this, especially since I
>
> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>
> although not as much).
>
>
> Both of the following conditions must exist:
>
> 1. You have at least one TCP local extension. In this case, due to Bria,
>
> etc. on an iPad, using TCP reduced battery usage since no need to run in
>
> background.
>
> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>
> seconds because it was automated at 2am each day. I needed it due to a long
>
> running sofia memory leak on OSX which may be resolved but I need to test.
>
>
> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060 port
>
> active (but NOT open) on the TCP stack, in my case it was 30 seconds but I
>
> found the time could be different on other platforms. When FreeSwitch comes
>
> back up the "internal.xml" profile fails because port 5060 is in use. If FS
>
> is brought up AFTER the TIME_WAIT goes away there is no problem, but in my
>
> case it was very quick each night to minimize the outage.
>
>
> Solution: I changed the OSX plist that auto restarts FS to run a script
> that
>
> starts FS, this makes no difference except a script is more flexible.
> Rather
>
> than use an arbitrary sleep time I check for port 5060 which is more
>
> reliable and faster. I tested this many times using different scenarios and
>
> all worked. Note I tried to use the "lsof -i TCP:5060" command but it shows
>
> only OPEN ports so it won't work.
>
>
> #!/bin/sh
>
> while : ; do
>
>    a=$(/usr/sbin/netstat -p tcp)
>
>    if [[ "$a" != *sip* ]] ; then
>
>      break
>
>    fi
>
>    echo "Port 5060 still in use - waiting for it to free up"
>
>    echo "${a}"
>
>    /bin/sleep 3
>
> done
>
> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>
>
> #   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
>
> #   if [ "$a" = "" ] ; then
>
>
>
>
> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>
>
> For anyone who may run into this see "FS shutdown not closing TCP port
>
> causes TIME_WAIT problems" for follow up and solution.
>
> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>
>
> That's what I was thinking, trying to figure out how now. Will post a
>
> solution when I get one.
>
>
> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>
>
> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>
>
> another MAC specific thing.
>
>
> I am not really sure what you can do since you tend to find issues
>
>
> deep inside libsofia-sip and its portability to OSX.
>
>
> Maybe make your restart script sleep a bit?
>
>
>
> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>
>
> su_setreuseaddr(socket, 1);
>
>
>
>
>
>
> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>
>
> If FS is shutdown and started too quickly, the internal profile ip/port
>
> stays in use and FS will fail to start it. After searching on the wen I
>
> found many references to TIME_WAIT but all fixes were to the program
> source.
>
> So I am wondering if when FS/sofia shuts down they are not releasing the
>
> ip/port for internal. If it's true, I found some source changes to solve
> it.
>
> Google "TIME-WAIT" and you'll find lots of good stuff like
>
>
> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait
> .
>
> Apparently there is no way to free up the port manually. Help/comments
>
> anyone?
>
>
> Mario G
>
>
> _________________________________________________________________________
>
>
>
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Wed Apr  4 01:10:29 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 3 Apr 2012 17:10:29 -0400
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
 causes TIME_WAIT problems
In-Reply-To: <CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
	<0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
	<CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>
Message-ID: <CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>

Even doing "fsctl shutdown restart" would restart freeswitch.

-BDF
On Apr 3, 2012 5:09 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> fs_cli -x "fsctl shutdown gracefully"
>
> -BDF
> On Apr 3, 2012 4:48 PM, "Mario G" <mario_fs at mgtech.com> wrote:
>
>> If you mean the effect, it did not stop FS although the results were the
>> same, it prevented the fast FS shutdown/startup from activating the
>> internal profile so *none* of the phone extensions work. In my case that
>> meant all phones were out from 2am to whenever I activated internal. Not
>> good in case of an emergency. Hope it may be useful to someone, it took me
>> 15 days to find and resolve.
>>
>> If you mean how is FreeSwitch shutdown, the 2am script runs a script that
>> runs one command: /usr/local/freeswitch/bin/fs_cli -x "shutdown"
>>
>> On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:
>>
>> How exactly does it "stop" freeswitch?
>>
>> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>>
>> I am posting this in case someone else runs into this, especially since I
>>
>> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>>
>> although not as much).
>>
>>
>> Both of the following conditions must exist:
>>
>> 1. You have at least one TCP local extension. In this case, due to Bria,
>>
>> etc. on an iPad, using TCP reduced battery usage since no need to run in
>>
>> background.
>>
>> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>>
>> seconds because it was automated at 2am each day. I needed it due to a
>> long
>>
>> running sofia memory leak on OSX which may be resolved but I need to test.
>>
>>
>> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060
>> port
>>
>> active (but NOT open) on the TCP stack, in my case it was 30 seconds but I
>>
>> found the time could be different on other platforms. When FreeSwitch
>> comes
>>
>> back up the "internal.xml" profile fails because port 5060 is in use. If
>> FS
>>
>> is brought up AFTER the TIME_WAIT goes away there is no problem, but in my
>>
>> case it was very quick each night to minimize the outage.
>>
>>
>> Solution: I changed the OSX plist that auto restarts FS to run a script
>> that
>>
>> starts FS, this makes no difference except a script is more flexible.
>> Rather
>>
>> than use an arbitrary sleep time I check for port 5060 which is more
>>
>> reliable and faster. I tested this many times using different scenarios
>> and
>>
>> all worked. Note I tried to use the "lsof -i TCP:5060" command but it
>> shows
>>
>> only OPEN ports so it won't work.
>>
>>
>> #!/bin/sh
>>
>> while : ; do
>>
>>    a=$(/usr/sbin/netstat -p tcp)
>>
>>    if [[ "$a" != *sip* ]] ; then
>>
>>      break
>>
>>    fi
>>
>>    echo "Port 5060 still in use - waiting for it to free up"
>>
>>    echo "${a}"
>>
>>    /bin/sleep 3
>>
>> done
>>
>> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>>
>>
>> #   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
>>
>> #   if [ "$a" = "" ] ; then
>>
>>
>>
>>
>> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>>
>>
>> For anyone who may run into this see "FS shutdown not closing TCP port
>>
>> causes TIME_WAIT problems" for follow up and solution.
>>
>> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>>
>>
>> That's what I was thinking, trying to figure out how now. Will post a
>>
>> solution when I get one.
>>
>>
>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>>
>>
>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>>
>>
>> another MAC specific thing.
>>
>>
>> I am not really sure what you can do since you tend to find issues
>>
>>
>> deep inside libsofia-sip and its portability to OSX.
>>
>>
>> Maybe make your restart script sleep a bit?
>>
>>
>>
>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>>
>>
>> su_setreuseaddr(socket, 1);
>>
>>
>>
>>
>>
>>
>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>
>>
>> If FS is shutdown and started too quickly, the internal profile ip/port
>>
>> stays in use and FS will fail to start it. After searching on the wen I
>>
>> found many references to TIME_WAIT but all fixes were to the program
>> source.
>>
>> So I am wondering if when FS/sofia shuts down they are not releasing the
>>
>> ip/port for internal. If it's true, I found some source changes to solve
>> it.
>>
>> Google "TIME-WAIT" and you'll find lots of good stuff like
>>
>>
>> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait
>> .
>>
>> Apparently there is no way to free up the port manually. Help/comments
>>
>> anyone?
>>
>>
>> Mario G
>>
>>
>> _________________________________________________________________________
>>
>>
>>
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Wed Apr  4 01:27:20 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 14:27:20 -0700
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
	causes TIME_WAIT problems
In-Reply-To: <CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
	<0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
	<CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>
	<CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>
Message-ID: <B80214C4-B8AF-4EB7-9B87-58B3FC790668@mgtech.com>

Doesn't solve the problem below since no matter how you restart the issue is there. FYI... Your command allows one "recycle" script to shutdown/start. But in my case not as convenient since OSX autostarts FS at system startup and anytime it goes down for any reason (such as a crash which it never does) so no need for the shutdown script to restart. Would be useful for other platforms though.
Mario G

On Apr 3, 2012, at 2:10 PM, Brian Foster wrote:

> Even doing "fsctl shutdown restart" would restart freeswitch.
> 
> -BDF
> 
> On Apr 3, 2012 5:09 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
> fs_cli -x "fsctl shutdown gracefully"
> 
> -BDF
> 
> On Apr 3, 2012 4:48 PM, "Mario G" <mario_fs at mgtech.com> wrote:
> If you mean the effect, it did not stop FS although the results were the same, it prevented the fast FS shutdown/startup from activating the internal profile so none of the phone extensions work. In my case that meant all phones were out from 2am to whenever I activated internal. Not good in case of an emergency. Hope it may be useful to someone, it took me 15 days to find and resolve.
> 
> If you mean how is FreeSwitch shutdown, the 2am script runs a script that runs one command: /usr/local/freeswitch/bin/fs_cli -x "shutdown"
> 
> On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:
> 
>> How exactly does it "stop" freeswitch?
>> 
>> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>>> I am posting this in case someone else runs into this, especially since I
>>> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>>> although not as much).
>>> 
>>> Both of the following conditions must exist:
>>> 1. You have at least one TCP local extension. In this case, due to Bria,
>>> etc. on an iPad, using TCP reduced battery usage since no need to run in
>>> background.
>>> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>>> seconds because it was automated at 2am each day. I needed it due to a long
>>> running sofia memory leak on OSX which may be resolved but I need to test.
>>> 
>>> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060 port
>>> active (but NOT open) on the TCP stack, in my case it was 30 seconds but I
>>> found the time could be different on other platforms. When FreeSwitch comes
>>> back up the "internal.xml" profile fails because port 5060 is in use. If FS
>>> is brought up AFTER the TIME_WAIT goes away there is no problem, but in my
>>> case it was very quick each night to minimize the outage.
>>> 
>>> Solution: I changed the OSX plist that auto restarts FS to run a script that
>>> starts FS, this makes no difference except a script is more flexible. Rather
>>> than use an arbitrary sleep time I check for port 5060 which is more
>>> reliable and faster. I tested this many times using different scenarios and
>>> all worked. Note I tried to use the "lsof -i TCP:5060" command but it shows
>>> only OPEN ports so it won't work.
>>> 
>>> #!/bin/sh
>>> while : ; do
>>>    a=$(/usr/sbin/netstat -p tcp)
>>>    if [[ "$a" != *sip* ]] ; then
>>>      break
>>>    fi
>>>    echo "Port 5060 still in use - waiting for it to free up"
>>>    echo "${a}"
>>>    /bin/sleep 3
>>> done
>>> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>>> 
>>> #   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
>>> #   if [ "$a" = "" ] ; then
>>> 
>>> 
>>> 
>>> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>>> 
>>> For anyone who may run into this see "FS shutdown not closing TCP port
>>> causes TIME_WAIT problems" for follow up and solution.
>>> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>>> 
>>> That's what I was thinking, trying to figure out how now. Will post a
>>> solution when I get one.
>>> 
>>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>>> 
>>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>>> 
>>> another MAC specific thing.
>>> 
>>> I am not really sure what you can do since you tend to find issues
>>> 
>>> deep inside libsofia-sip and its portability to OSX.
>>> 
>>> Maybe make your restart script sleep a bit?
>>> 
>>> 
>>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>>> 
>>> su_setreuseaddr(socket, 1);
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>> 
>>> If FS is shutdown and started too quickly, the internal profile ip/port
>>> stays in use and FS will fail to start it. After searching on the wen I
>>> found many references to TIME_WAIT but all fixes were to the program source.
>>> So I am wondering if when FS/sofia shuts down they are not releasing the
>>> ip/port for internal. If it's true, I found some source changes to solve it.
>>> Google "TIME-WAIT" and you'll find lots of good stuff like
>>> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
>>> Apparently there is no way to free up the port manually. Help/comments
>>> anyone?
>>> 
>>> Mario G
>>> 
>>> _________________________________________________________________________
>>> 
>>> 
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Wed Apr  4 01:28:20 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 3 Apr 2012 16:28:20 -0500
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
 causes TIME_WAIT problems
In-Reply-To: <CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
	<0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
	<CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>
	<CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>
Message-ID: <CAKbxfG_krHJc0R22EFhb8Mx5JceyQ=m-B_KKFrmmcvDkzxNVCA@mail.gmail.com>

I meant the command it uses.  One guess is that since you are using
fs_cli the shutdown probably kicks you off the event socket mid
shutdown while the rest of FS is still shutting down and it tries to
restart it again too soon.

you could try fs_cli -x "fsctl shutdown now"
this will instantly end FS

otherwise maybe its just a "mac thing" as suspected.


On Tue, Apr 3, 2012 at 4:10 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> Even doing "fsctl shutdown restart" would restart freeswitch.
>
> -BDF
>
> On Apr 3, 2012 5:09 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>
>> fs_cli -x "fsctl shutdown gracefully"
>>
>> -BDF
>>
>> On Apr 3, 2012 4:48 PM, "Mario G" <mario_fs at mgtech.com> wrote:
>>>
>>> If you mean the effect, it did not stop FS although the results were the
>>> same, it prevented the fast FS shutdown/startup from activating the internal
>>> profile so none of the phone extensions work. In my case that meant all
>>> phones were out from 2am to whenever I activated internal. Not good in case
>>> of an emergency. Hope it may be useful to someone, it took me 15 days to
>>> find and resolve.
>>>
>>> If you mean how is FreeSwitch shutdown, the 2am script runs a script that
>>> runs one command:?/usr/local/freeswitch/bin/fs_cli -x "shutdown"
>>>
>>> On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:
>>>
>>> How exactly does it "stop" freeswitch?
>>>
>>> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>> I am posting this in case someone else runs into this, especially since I
>>>
>>> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>>>
>>> although not as much).
>>>
>>>
>>> Both of the following conditions must exist:
>>>
>>> 1. You have at least one TCP local extension. In this case, due to Bria,
>>>
>>> etc. on an iPad, using TCP reduced battery usage since no need to run in
>>>
>>> background.
>>>
>>> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>>>
>>> seconds because it was automated at 2am each day. I needed it due to a
>>> long
>>>
>>> running sofia memory leak on OSX which may be resolved but I need to
>>> test.
>>>
>>>
>>> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060
>>> port
>>>
>>> active (but NOT open) on the TCP stack, in my case it was 30 seconds but
>>> I
>>>
>>> found the time could be different on other platforms. When FreeSwitch
>>> comes
>>>
>>> back up the "internal.xml" profile fails because port 5060 is in use. If
>>> FS
>>>
>>> is brought up AFTER the TIME_WAIT goes away there is no problem, but in
>>> my
>>>
>>> case it was very quick each night to minimize the outage.
>>>
>>>
>>> Solution: I changed the OSX plist that auto restarts FS to run a script
>>> that
>>>
>>> starts FS, this makes no difference except a script is more flexible.
>>> Rather
>>>
>>> than use an arbitrary sleep time I check for port 5060 which is more
>>>
>>> reliable and faster. I tested this many times using different scenarios
>>> and
>>>
>>> all worked. Note I tried to use the "lsof -i TCP:5060" command but it
>>> shows
>>>
>>> only OPEN ports so it won't work.
>>>
>>>
>>> #!/bin/sh
>>>
>>> while : ; do
>>>
>>> ?? a=$(/usr/sbin/netstat -p tcp)
>>>
>>> ?? if [[ "$a" != *sip* ]] ; then
>>>
>>> ?? ? break
>>>
>>> ?? fi
>>>
>>> ?? echo "Port 5060 still in use - waiting for it to free up"
>>>
>>> ?? echo "${a}"
>>>
>>> ?? /bin/sleep 3
>>>
>>> done
>>>
>>> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>>>
>>>
>>> # ? a=$(/usr/sbin/lsof -i TCP:5060)? <- only shows open ports
>>>
>>> # ? if [ "$a" = "" ] ; then
>>>
>>>
>>>
>>>
>>> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>>>
>>>
>>> For anyone who may run into this see "FS shutdown not closing TCP port
>>>
>>> causes TIME_WAIT problems" for follow up and solution.
>>>
>>> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>>>
>>>
>>> That's what I was thinking, trying to figure out how now. Will post a
>>>
>>> solution when I get one.
>>>
>>>
>>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>>>
>>>
>>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>>>
>>>
>>> another MAC specific thing.
>>>
>>>
>>> I am not really sure what you can do since you tend to find issues
>>>
>>>
>>> deep inside libsofia-sip and its portability to OSX.
>>>
>>>
>>> Maybe make your restart script sleep a bit?
>>>
>>>
>>>
>>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>>>
>>>
>>> su_setreuseaddr(socket, 1);
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>
>>>
>>> If FS is shutdown and started too quickly, the internal profile ip/port
>>>
>>> stays in use and FS will fail to start it. After searching on the wen I
>>>
>>> found many references to TIME_WAIT but all fixes were to the program
>>> source.
>>>
>>> So I am wondering if when FS/sofia shuts down they are not releasing the
>>>
>>> ip/port for internal. If it's true, I found some source changes to solve
>>> it.
>>>
>>> Google "TIME-WAIT" and you'll find lots of good stuff like
>>>
>>>
>>> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
>>>
>>> Apparently there is no way to free up the port manually. Help/comments
>>>
>>> anyone?
>>>
>>>
>>> Mario G
>>>
>>>
>>> _________________________________________________________________________
>>>
>>>
>>>
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mario_fs at mgtech.com  Wed Apr  4 02:37:47 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 3 Apr 2012 15:37:47 -0700
Subject: [Freeswitch-users] [Solved] FS shutdown not closing TCP port
	causes TIME_WAIT problems
In-Reply-To: <CAKbxfG_krHJc0R22EFhb8Mx5JceyQ=m-B_KKFrmmcvDkzxNVCA@mail.gmail.com>
References: <B99ED5C7-B0B7-47AB-8290-383BA0F2ACAA@mgtech.com>
	<CAKbxfG_ZzxZ1tr=AwSv8S3dM3h+EV79F=6gba83_diVMTULvCA@mail.gmail.com>
	<E05DB5F5-33C2-4C70-8CEF-045339C1D8E7@mgtech.com>
	<C94D9124-6384-4536-9749-8C8CD9E75ADC@mgtech.com>
	<350366AB-F14F-4411-9A79-434EC599A80D@mgtech.com>
	<CAKbxfG9sbJ8dQKEmdVgXcg0WonMSu3FDonc4SfvkHdHneBPGeQ@mail.gmail.com>
	<0239B1C7-8DA0-418C-9024-116F2BAC9504@mgtech.com>
	<CAKV-jFwmezh8U-YL1wGTVsqS822amnaOrRBYWysOswNa--zLiA@mail.gmail.com>
	<CAKV-jFzv6_SQ_jjiPWGTFrZ7Bm2e9CjbSQJpjuXWX=-Bkdnxxg@mail.gmail.com>
	<CAKbxfG_krHJc0R22EFhb8Mx5JceyQ=m-B_KKFrmmcvDkzxNVCA@mail.gmail.com>
Message-ID: <B7A7E577-F8CF-484E-B1C1-1993B8EDDBE0@mgtech.com>

Happy to report (for my sake) that using the exact command below (fs_cli in Terminal) did not solve the problem (I tested 4 times), ports went to TIME_WAIT. Good idea though...  I was able to duplicate it intermittently on OpenSUSE but consistently on OSX. I will add this tidbit to the OSX wiki when I update it, hopefully soon. Thanks for the suggestion.
Mario G

On Apr 3, 2012, at 2:28 PM, Anthony Minessale wrote:

> I meant the command it uses.  One guess is that since you are using
> fs_cli the shutdown probably kicks you off the event socket mid
> shutdown while the rest of FS is still shutting down and it tries to
> restart it again too soon.
> 
> you could try fs_cli -x "fsctl shutdown now"
> this will instantly end FS
> 
> otherwise maybe its just a "mac thing" as suspected.
> 
> 
> On Tue, Apr 3, 2012 at 4:10 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> Even doing "fsctl shutdown restart" would restart freeswitch.
>> 
>> -BDF
>> 
>> On Apr 3, 2012 5:09 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>> 
>>> fs_cli -x "fsctl shutdown gracefully"
>>> 
>>> -BDF
>>> 
>>> On Apr 3, 2012 4:48 PM, "Mario G" <mario_fs at mgtech.com> wrote:
>>>> 
>>>> If you mean the effect, it did not stop FS although the results were the
>>>> same, it prevented the fast FS shutdown/startup from activating the internal
>>>> profile so none of the phone extensions work. In my case that meant all
>>>> phones were out from 2am to whenever I activated internal. Not good in case
>>>> of an emergency. Hope it may be useful to someone, it took me 15 days to
>>>> find and resolve.
>>>> 
>>>> If you mean how is FreeSwitch shutdown, the 2am script runs a script that
>>>> runs one command: /usr/local/freeswitch/bin/fs_cli -x "shutdown"
>>>> 
>>>> On Apr 3, 2012, at 11:24 AM, Anthony Minessale wrote:
>>>> 
>>>> How exactly does it "stop" freeswitch?
>>>> 
>>>> On Tue, Apr 3, 2012 at 11:17 AM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> I am posting this in case someone else runs into this, especially since I
>>>> 
>>>> found it may not be Mac OSX specific (I was able to duplicate on SUSE
>>>> 
>>>> although not as much).
>>>> 
>>>> 
>>>> Both of the following conditions must exist:
>>>> 
>>>> 1. You have at least one TCP local extension. In this case, due to Bria,
>>>> 
>>>> etc. on an iPad, using TCP reduced battery usage since no need to run in
>>>> 
>>>> background.
>>>> 
>>>> 2. FreeSwitch is recycled very quickly, in this case down/up within 30
>>>> 
>>>> seconds because it was automated at 2am each day. I needed it due to a
>>>> long
>>>> 
>>>> running sofia memory leak on OSX which may be resolved but I need to
>>>> test.
>>>> 
>>>> 
>>>> Problem: The extension using TCP cause a TIME_WAIT that keeps the 5060
>>>> port
>>>> 
>>>> active (but NOT open) on the TCP stack, in my case it was 30 seconds but
>>>> I
>>>> 
>>>> found the time could be different on other platforms. When FreeSwitch
>>>> comes
>>>> 
>>>> back up the "internal.xml" profile fails because port 5060 is in use. If
>>>> FS
>>>> 
>>>> is brought up AFTER the TIME_WAIT goes away there is no problem, but in
>>>> my
>>>> 
>>>> case it was very quick each night to minimize the outage.
>>>> 
>>>> 
>>>> Solution: I changed the OSX plist that auto restarts FS to run a script
>>>> that
>>>> 
>>>> starts FS, this makes no difference except a script is more flexible.
>>>> Rather
>>>> 
>>>> than use an arbitrary sleep time I check for port 5060 which is more
>>>> 
>>>> reliable and faster. I tested this many times using different scenarios
>>>> and
>>>> 
>>>> all worked. Note I tried to use the "lsof -i TCP:5060" command but it
>>>> shows
>>>> 
>>>> only OPEN ports so it won't work.
>>>> 
>>>> 
>>>> #!/bin/sh
>>>> 
>>>> while : ; do
>>>> 
>>>>    a=$(/usr/sbin/netstat -p tcp)
>>>> 
>>>>    if [[ "$a" != *sip* ]] ; then
>>>> 
>>>>      break
>>>> 
>>>>    fi
>>>> 
>>>>    echo "Port 5060 still in use - waiting for it to free up"
>>>> 
>>>>    echo "${a}"
>>>> 
>>>>    /bin/sleep 3
>>>> 
>>>> done
>>>> 
>>>> /usr/local/freeswitch/bin/freeswitch -nc -nonat
>>>> 
>>>> 
>>>> #   a=$(/usr/sbin/lsof -i TCP:5060)  <- only shows open ports
>>>> 
>>>> #   if [ "$a" = "" ] ; then
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Apr 3, 2012, at 8:53 AM, Mario G wrote:
>>>> 
>>>> 
>>>> For anyone who may run into this see "FS shutdown not closing TCP port
>>>> 
>>>> causes TIME_WAIT problems" for follow up and solution.
>>>> 
>>>> On Apr 2, 2012, at 1:33 PM, Mario G wrote:
>>>> 
>>>> 
>>>> That's what I was thinking, trying to figure out how now. Will post a
>>>> 
>>>> solution when I get one.
>>>> 
>>>> 
>>>> On Apr 2, 2012, at 1:28 PM, Anthony Minessale wrote:
>>>> 
>>>> 
>>>> From what I see, sofia uses the REUSEADDR flag so I will bet its yet
>>>> 
>>>> 
>>>> another MAC specific thing.
>>>> 
>>>> 
>>>> I am not really sure what you can do since you tend to find issues
>>>> 
>>>> 
>>>> deep inside libsofia-sip and its portability to OSX.
>>>> 
>>>> 
>>>> Maybe make your restart script sleep a bit?
>>>> 
>>>> 
>>>> 
>>>> libs/sofia-sip/libsofia-sip-ua/tport/tport_type_tcp.c:
>>>> 
>>>> 
>>>> su_setreuseaddr(socket, 1);
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Mon, Apr 2, 2012 at 3:13 PM, Mario G <mario_fs at mgtech.com> wrote:
>>>> 
>>>> 
>>>> If FS is shutdown and started too quickly, the internal profile ip/port
>>>> 
>>>> stays in use and FS will fail to start it. After searching on the wen I
>>>> 
>>>> found many references to TIME_WAIT but all fixes were to the program
>>>> source.
>>>> 
>>>> So I am wondering if when FS/sofia shuts down they are not releasing the
>>>> 
>>>> ip/port for internal. If it's true, I found some source changes to solve
>>>> it.
>>>> 
>>>> Google "TIME-WAIT" and you'll find lots of good stuff like
>>>> 
>>>> 
>>>> http://serverfault.com/questions/329845/how-to-forcibly-close-a-socket-in-time-wait.
>>>> 
>>>> Apparently there is no way to free up the port manually. Help/comments
>>>> 
>>>> anyone?
>>>> 
>>>> 
>>>> Mario G
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> 
>>>> 
>>>> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bote_radio at botecomm.com  Wed Apr  4 08:03:14 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 4 Apr 2012 00:03:14 -0400
Subject: [Freeswitch-users] adding leg C for external recorder
Message-ID: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>

What are the ways that I can set up an outbound call to send an additional
leg to an external logging recorder?

Right now I'm trying to use mod_conference, but it just ends the call as
soon as it begins, even though I have another dialplan extension that sets
up several endpoints in a conference bridge just fine.

I thought about using eavesdrop, but that is initiated by and sends audio
to leg A. Plus it joins an existing call whereas in this application I'm
establishing a new call, so that won't work. I need to send the audio to
leg C from the start.

I'm hoping I don't have to write a script, because I'm on the customer's
site and I'm running out of time.

Any ideas? Thanks.

Bote
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From sardor.tukhtaev at gmail.com  Wed Apr  4 08:53:32 2012
From: sardor.tukhtaev at gmail.com (sardor)
Date: Tue, 3 Apr 2012 21:53:32 -0700 (PDT)
Subject: [Freeswitch-users] Configurations external and intenal calls in
	FreeSwitch
Message-ID: <1333515212323-7435325.post@n2.nabble.com>

Hello everyone,

Could you help me to confgure out my *.xml files for receiving calls to SIP
and making external calls to regular phones. I've done a lot of changes in
this side, but no success. Please if you can advise smth or send copy of xml
files, then it would be great help for me. 
Thanks in advance...

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configurations-external-and-intenal-calls-in-FreeSwitch-tp7435325p7435325.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From krice at freeswitch.org  Wed Apr  4 09:27:29 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 04 Apr 2012 00:27:29 -0500
Subject: [Freeswitch-users] Fisheye and Jira Upgraded!
Message-ID: <CBA145F1.30D19%krice@freeswitch.org>

Just a quick announcement.

As you may have noticed end of the Day US Time on Tuesday April 3rd, Fisheye
and Jira were offline for a while. However, I am pleased to report that both
have now been upgraded, and a Hardware replacement has happened on both
hosts.

Please report any issues to me or BKW via the mailing list or join us on the
IRC Channel at #FreeSWITCH on irc.freenode.net

Thanks for your Support!

K
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From anton.jugatsu at gmail.com  Wed Apr  4 10:02:36 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 4 Apr 2012 10:02:36 +0400
Subject: [Freeswitch-users] Configurations external and intenal calls in
	FreeSwitch
In-Reply-To: <1333515212323-7435325.post@n2.nabble.com>
References: <1333515212323-7435325.post@n2.nabble.com>
Message-ID: <CANBhyPAiGgJ-GXueeNTFbwWbpqeZwOyd6CMYS0trVC9w31_7nA@mail.gmail.com>

Check out wiki.freeswitch.org for VOIP-providers configuration. Also, you
didn't paste any logs and configs.

4 ?????? 2012 ?. 8:53 ???????????? sardor <sardor.tukhtaev at gmail.com>???????:

> Hello everyone,
>
> Could you help me to confgure out my *.xml files for receiving calls to SIP
> and making external calls to regular phones. I've done a lot of changes in
> this side, but no success. Please if you can advise smth or send copy of
> xml
> files, then it would be great help for me.
> Thanks in advance...
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Configurations-external-and-intenal-calls-in-FreeSwitch-tp7435325p7435325.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Wed Apr  4 10:05:42 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 4 Apr 2012 09:05:42 +0300
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
Message-ID: <CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>

Use a media bug to record the call?
API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
Dialplan:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session

Or did you mean something completely different?

-Avi


On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com> wrote:

> What are the ways that I can set up an outbound call to send an additional
> leg to an external logging recorder?
>
> Right now I'm trying to use mod_conference, but it just ends the call as
> soon as it begins, even though I have another dialplan extension that sets
> up several endpoints in a conference bridge just fine.
>
> I thought about using eavesdrop, but that is initiated by and sends audio
> to leg A. Plus it joins an existing call whereas in this application I'm
> establishing a new call, so that won't work. I need to send the audio to
> leg C from the start.
>
> I'm hoping I don't have to write a script, because I'm on the customer's
> site and I'm running out of time.
>
> Any ideas? Thanks.
>
> Bote
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bote_radio at botecomm.com  Wed Apr  4 10:32:23 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 4 Apr 2012 02:32:23 -0400
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
Message-ID: <CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>

It's a ridiculously expensive stand-alone recording system that can record
RTP streams. It's not up to me to change it at this point.

My leg C notion is a fallback in case it doesn't grok SIP calls as
advertised; everything else in this project has fallen short of
expectations (except FreeSWITCH) so I expect this to fall short as well.
:- /

Bote



On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Use a media bug to record the call?
> API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
> Dialplan:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>
> Or did you mean something completely different?
>
> -Avi
>
>
> On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com> wrote:
>
>> What are the ways that I can set up an outbound call to send an
>> additional leg to an external logging recorder?
>>
>> Right now I'm trying to use mod_conference, but it just ends the call as
>> soon as it begins, even though I have another dialplan extension that sets
>> up several endpoints in a conference bridge just fine.
>>
>> I thought about using eavesdrop, but that is initiated by and sends audio
>> to leg A. Plus it joins an existing call whereas in this application I'm
>> establishing a new call, so that won't work. I need to send the audio to
>> leg C from the start.
>>
>> I'm hoping I don't have to write a script, because I'm on the customer's
>> site and I'm running out of time.
>>
>> Any ideas? Thanks.
>>
>> Bote
>>
>>
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From avi at avimarcus.net  Wed Apr  4 10:48:29 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 4 Apr 2012 09:48:29 +0300
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
	<CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
Message-ID: <CALNZuwnmRhVEeyWoQk+J04CCV9fnsGm046F8A9HjMr6q+D19Rw@mail.gmail.com>

Usually an RTP stream recorder records RTP streams that pass through its
network. Can you jigger your network to have it pass through that box?
Or some other local packet capture should be able to grab and then forward
to another machine.

Also, mod_conference should work if the logger picks up the call..

What features does this external solution offer you, to make it worth the
effort?

-Avi


On Wed, Apr 4, 2012 at 9:32 AM, Bote Man <bote_radio at botecomm.com> wrote:

> It's a ridiculously expensive stand-alone recording system that can record
> RTP streams. It's not up to me to change it at this point.
>
> My leg C notion is a fallback in case it doesn't grok SIP calls as
> advertised; everything else in this project has fallen short of
> expectations (except FreeSWITCH) so I expect this to fall short as well.
> :- /
>
> Bote
>
>
>
>
> On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Use a media bug to record the call?
>> API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
>> Dialplan:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>>
>> Or did you mean something completely different?
>>
>> -Avi
>>
>>
>> On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com> wrote:
>>
>>> What are the ways that I can set up an outbound call to send an
>>> additional leg to an external logging recorder?
>>>
>>> Right now I'm trying to use mod_conference, but it just ends the call as
>>> soon as it begins, even though I have another dialplan extension that sets
>>> up several endpoints in a conference bridge just fine.
>>>
>>> I thought about using eavesdrop, but that is initiated by and sends
>>> audio to leg A. Plus it joins an existing call whereas in this application
>>> I'm establishing a new call, so that won't work. I need to send the audio
>>> to leg C from the start.
>>>
>>> I'm hoping I don't have to write a script, because I'm on the customer's
>>> site and I'm running out of time.
>>>
>>> Any ideas? Thanks.
>>>
>>> Bote
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gcd at i.ph  Wed Apr  4 11:13:56 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 4 Apr 2012 15:13:56 +0800
Subject: [Freeswitch-users] Configurations external and intenal calls in
	FreeSwitch
In-Reply-To: <1333515212323-7435325.post@n2.nabble.com>
References: <1333515212323-7435325.post@n2.nabble.com>
Message-ID: <CAHKqMb7S4hHpZmbSGEFKAdNjwUD=XYNfZat9mJ3Zu+T9yM39qA@mail.gmail.com>

also include details of your PSTN hardware.

On Wed, Apr 4, 2012 at 12:53 PM, sardor <sardor.tukhtaev at gmail.com> wrote:

> Hello everyone,
>
> Could you help me to confgure out my *.xml files for receiving calls to SIP
> and making external calls to regular phones. I've done a lot of changes in
> this side, but no success. Please if you can advise smth or send copy of
> xml
> files, then it would be great help for me.
> Thanks in advance...
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Configurations-external-and-intenal-calls-in-FreeSwitch-tp7435325p7435325.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sardor.tukhtaev at gmail.com  Wed Apr  4 11:49:35 2012
From: sardor.tukhtaev at gmail.com (sardor)
Date: Wed, 4 Apr 2012 00:49:35 -0700 (PDT)
Subject: [Freeswitch-users] Configurations external and intenal calls in
	FreeSwitch
In-Reply-To: <CAHKqMb7S4hHpZmbSGEFKAdNjwUD=XYNfZat9mJ3Zu+T9yM39qA@mail.gmail.com>
References: <1333515212323-7435325.post@n2.nabble.com>
	<CAHKqMb7S4hHpZmbSGEFKAdNjwUD=XYNfZat9mJ3Zu+T9yM39qA@mail.gmail.com>
Message-ID: <CAPu2oOL7grsngpvgtQtLRsKF8n3POb2BJoucpRi3ZRwMSv-_tQ@mail.gmail.com>

It must be Cisco's hardware.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Configurations-external-and-intenal-calls-in-FreeSwitch-tp7435325p7435690.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From brett at launch3.net  Wed Apr  4 12:06:12 2012
From: brett at launch3.net (Brett Wilson)
Date: Wed, 4 Apr 2012 04:06:12 -0400
Subject: [Freeswitch-users] double nat trouble
Message-ID: <00f501cd1239$ccfd7190$66f854b0$@launch3.net>

Hey guys,

We have just temporarily moved our office, so our phones are no longer just
on the office LAN. They are on a remote lan, both sides behind NAT and I'm
trying to get them working right but I am having trouble. I have things
working to the point where I can call to the POTS or to an extension on the
local network back home just fine. The problem is that the remote phones do
not receive sip messages behind the nat. It seems to me that freeswitch is
sending the messages to the sip port the phones are configured with, and not
the NAT port that can be seen when viewing the registrations status. The nat
ports are very high numbers. I suspect if FS was to send messages to those
ports, it would get through the foreign nat and reach the phone.

 

On the FS side, I am using pfsense as the firewall, and have port forwards
set up to the fs box. I am using a dynamic ip with dyn-dns-like service and
am using that domain name in the fs internal/external profiles for the
ext-sip-ip and ext-rtp-ip variables. I have upnp turned off but it does not
make any difference anyway.

 

Strangely, I was actually able to place a call from one remote phone to
another once earlier, but it must have been a fluke because I haven't had it
working after that.

 

What am I missing here?

 

******************************************* 
Brett Wilson
IT Department
Launch 3 Ventures, LLC 
134 Myer Street 
Hackensack, NJ 07601 
Phone: 877.878.9134
Fax: 646.536.3866 
Email: Brett.Wilson at launch3.net
AOL IM: Brett.Wilson at launch3.net
 <http://www.launch3.net/> www.Launch3.net 
 <http://www.launch3telecom.com/> www.Launch3telecom.com 
******************************************* 
 <http://launch3telecom.blogspot.com/> Description: Description:
Blogger-logo
<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>
Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom>
Description: Description: Twitter-Logo
<https://plus.google.com/102713212642017290959> Description: Description:
GPlus-Logo

 

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From Daniel.Knaggs at realitysolutions.co.uk  Wed Apr  4 13:52:14 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Wed, 4 Apr 2012 09:52:14 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05A8C2@RSV16.realitysolutions.local>

OK, it?s now tracing on the entire span.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 03 April 2012 18:24
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

Oops, mea culpa! I meant "ftdm trace" which will record the audio on a whole span or a single channel depending on the args.
-MC
On Tue, Apr 3, 2012 at 8:52 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Audio was captured using ?record_session? where it didn?t appear in the audio by ?ear?.

?ftdm dump? displays details about the span I tell it to: -

span_id: 1
chan_id: 1
physical_span_id: 1
physical_chan_id: 1
physical_status: ok
physical_status_red: 0
physical_status_yellow: 0
physical_status_rai: 0
physical_status_blue: 0
physical_status_ais: 0
physical_status_general: 0
signaling_status: UP
type: B
state: DOWN
last_state: HANGUP_COMPLETE
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)

etc?



[cid:imagece0f1a.PNG at 964b0c37.4ba96b9a]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:image560aed.PNG at 0dbfbf14.49bc30d0]





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From david.villasmil.work at gmail.com  Wed Apr  4 14:26:51 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 4 Apr 2012 12:26:51 +0200
Subject: [Freeswitch-users] FreeSWITCH-Billing
In-Reply-To: <CAFGRPVof4tLBSakAv_VBFY8ptufL5OSF8Nrh0QfexJ19aV2KaQ@mail.gmail.com>
References: <CAKMOEZkEinT7FBLQQuMS2x9tuGs8E5NmgCDKQGnvs-JvQO+cLw@mail.gmail.com>
	<CAFGRPVof4tLBSakAv_VBFY8ptufL5OSF8Nrh0QfexJ19aV2KaQ@mail.gmail.com>
Message-ID: <CAFGRPVrCb9Hd+pti4Wou__ezQC1MEWHqCMCiQ0h-dQcFBCOowA@mail.gmail.com>

Also, remember to set the correct info in the "variables.conf" file.

David

On Wed, Apr 4, 2012 at 12:26 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Dihn,
>
> There is a script called "install.pl", you need to execute that.
>
> David
>
>
> On Wed, Apr 4, 2012 at 7:02 AM, dinh thuc <dinhthuc1303 at gmail.com> wrote:
>
>> Hi David !!
>>
>> We need to solution about billing in my system. I have read link
>> https://github.com/davidcsi/FreeSWITCH-Billing
>>
>> You have script for install all ?
>>
>> I want to install and test
>>
>> B&R
>>
>> Dinh Thuc
>>
>
>
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From wesleyakio at tuntscorp.com  Wed Apr  4 15:01:12 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Wed, 4 Apr 2012 08:01:12 -0300
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CALNZuwnmRhVEeyWoQk+J04CCV9fnsGm046F8A9HjMr6q+D19Rw@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
	<CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
	<CALNZuwnmRhVEeyWoQk+J04CCV9fnsGm046F8A9HjMr6q+D19Rw@mail.gmail.com>
Message-ID: <CAGTsDSuvBauq71FVVZ8MTeN4B6=YhBc7cBuQMfqxbvfTjmvrTQ@mail.gmail.com>

I guess mod_spy will do the trick for you... Take a look at the wiki page
and let us know...

Best,

Wesley Akio
Em 04/04/2012 03:49, "Avi Marcus" <avi at avimarcus.net> escreveu:

> Usually an RTP stream recorder records RTP streams that pass through its
> network. Can you jigger your network to have it pass through that box?
> Or some other local packet capture should be able to grab and then forward
> to another machine.
>
> Also, mod_conference should work if the logger picks up the call..
>
> What features does this external solution offer you, to make it worth the
> effort?
>
> -Avi
>
>
> On Wed, Apr 4, 2012 at 9:32 AM, Bote Man <bote_radio at botecomm.com> wrote:
>
>> It's a ridiculously expensive stand-alone recording system that can
>> record RTP streams. It's not up to me to change it at this point.
>>
>> My leg C notion is a fallback in case it doesn't grok SIP calls as
>> advertised; everything else in this project has fallen short of
>> expectations (except FreeSWITCH) so I expect this to fall short as well.
>> :- /
>>
>> Bote
>>
>>
>>
>>
>> On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> Use a media bug to record the call?
>>> API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
>>> Dialplan:
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>>>
>>> Or did you mean something completely different?
>>>
>>> -Avi
>>>
>>>
>>> On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com>wrote:
>>>
>>>> What are the ways that I can set up an outbound call to send an
>>>> additional leg to an external logging recorder?
>>>>
>>>> Right now I'm trying to use mod_conference, but it just ends the call
>>>> as soon as it begins, even though I have another dialplan extension that
>>>> sets up several endpoints in a conference bridge just fine.
>>>>
>>>> I thought about using eavesdrop, but that is initiated by and sends
>>>> audio to leg A. Plus it joins an existing call whereas in this application
>>>> I'm establishing a new call, so that won't work. I need to send the audio
>>>> to leg C from the start.
>>>>
>>>> I'm hoping I don't have to write a script, because I'm on the
>>>> customer's site and I'm running out of time.
>>>>
>>>> Any ideas? Thanks.
>>>>
>>>> Bote
>>>>
>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From dgarcia at anew.com.ve  Wed Apr  4 15:20:01 2012
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Wed, 04 Apr 2012 06:50:01 -0430
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
	<CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
Message-ID: <4F7C2E61.2050401@anew.com.ve>

You could do this doing it at router level. This kind of recorders just 
capture rtp streams but not get involved in the call flow. So if you 
router have the feature, just pick a port and reproduce all traffic in 
that port then connect it to your recorder. In that way, your recorder 
will see all rtp packet in your netwokr


On 4/4/2012 2:02 AM, Bote Man wrote:
> It's a ridiculously expensive stand-alone recording system that can 
> record RTP streams. It's not up to me to change it at this point.
>
> My leg C notion is a fallback in case it doesn't grok SIP calls as 
> advertised; everything else in this project has fallen short of 
> expectations (except FreeSWITCH) so I expect this to fall short as 
> well.  :- /
>
> Bote
>
>
>
> On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi at avimarcus.net 
> <mailto:avi at avimarcus.net>> wrote:
>
>     Use a media bug to record the call?
>     API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
>     Dialplan:
>     http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>
>     Or did you mean something completely different?
>
>     -Avi
>
>
>     On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com
>     <mailto:bote_radio at botecomm.com>> wrote:
>
>         What are the ways that I can set up an outbound call to send
>         an additional leg to an external logging recorder?
>
>         Right now I'm trying to use mod_conference, but it just ends
>         the call as soon as it begins, even though I have another
>         dialplan extension that sets up several endpoints in a
>         conference bridge just fine.
>
>         I thought about using eavesdrop, but that is initiated by and
>         sends audio to leg A. Plus it joins an existing call whereas
>         in this application I'm establishing a new call, so that won't
>         work. I need to send the audio to leg C from the start.
>
>         I'm hoping I don't have to write a script, because I'm on the
>         customer's site and I'm running out of time.
>
>         Any ideas? Thanks.
>
>         Bote
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.1913 / Virus Database: 2114/4913 - Release Date: 04/03/12
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve

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From ali.jawad at splendor.net  Wed Apr  4 15:36:21 2012
From: ali.jawad at splendor.net (Ali Jawad)
Date: Wed, 4 Apr 2012 14:36:21 +0300
Subject: [Freeswitch-users] Handing calls coming from Kamailio
Message-ID: <CAG2QdgbgpQ4mnnaPfGk1zvzAjVWyP7G5_-B5TpmXd68Cy588-A@mail.gmail.com>

Hi
I have the following setup

sip phone --- Kamailio for Registration and Routing - Freeswitch for
billing and media handling - VOIP Provider

Calls are sent from Kamailio to FS and Authentication works as it should,
but I cant get the calls to be forwarded to the VOIP provider when they
arrive from Kamailio, if I register directly on FS the calls are delivered
to the VOIP provider and I can make a call, I did try to create an inbound
route but that did not work.

Any hints on what I have missed please ?

Thanks
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From bote_radio at botecomm.com  Wed Apr  4 16:53:53 2012
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 4 Apr 2012 08:53:53 -0400
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <4F7C2E61.2050401@anew.com.ve>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
	<CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
	<4F7C2E61.2050401@anew.com.ve>
Message-ID: <CAKYdwTbcfx0Up2ZKVpSKfGxx4FbwJ1JzLArwh7_zenREGRn_+g@mail.gmail.com>

Yes, I am hoping that this Cisco supports SPAN (Switched Port Analyzer)
because that is exactly what I need. Unfortunately, that is a job for
someone else involved in this project due to political and business
reasons, so I can not implement it or even check if it has that feature.

The problem with eavesdrop and mod_spy is that they depend on the
listener/recorder initiating the dialplan action. The recorder is passive
so the stream must be sent to it as another party to the call, unless I am
missing something here.

Thanks.

Bote


On Wed, Apr 4, 2012 at 7:20 AM, Saugort Dario Garcia Tovar <
dgarcia at anew.com.ve> wrote:

>  You could do this doing it at router level. This kind of recorders just
> capture rtp streams but not get involved in the call flow. So if you router
> have the feature, just pick a port and reproduce all traffic in that port
> then connect it to your recorder. In that way, your recorder will see all
> rtp packet in your netwokr
>
>
>
> On 4/4/2012 2:02 AM, Bote Man wrote:
>
> It's a ridiculously expensive stand-alone recording system that can record
> RTP streams. It's not up to me to change it at this point.
>
> My leg C notion is a fallback in case it doesn't grok SIP calls as
> advertised; everything else in this project has fallen short of
> expectations (except FreeSWITCH) so I expect this to fall short as well.
> :- /
>
> Bote
>
>
>
> On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Use a media bug to record the call?
>> API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record
>> Dialplan:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
>>
>>  Or did you mean something completely different?
>>
>> -Avi
>>
>>
>>  On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio at botecomm.com>wrote:
>>
>>>  What are the ways that I can set up an outbound call to send an
>>> additional leg to an external logging recorder?
>>>
>>> Right now I'm trying to use mod_conference, but it just ends the call as
>>> soon as it begins, even though I have another dialplan extension that sets
>>> up several endpoints in a conference bridge just fine.
>>>
>>> I thought about using eavesdrop, but that is initiated by and sends
>>> audio to leg A. Plus it joins an existing call whereas in this application
>>> I'm establishing a new call, so that won't work. I need to send the audio
>>> to leg C from the start.
>>>
>>> I'm hoping I don't have to write a script, because I'm on the customer's
>>> site and I'm running out of time.
>>>
>>> Any ideas? Thanks.
>>>
>>> Bote
>>>
>>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2012.0.1913 / Virus Database: 2114/4913 - Release Date: 04/03/12
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gerry at pstn2.net  Wed Apr  4 17:04:51 2012
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 4 Apr 2012 09:04:51 -0400
Subject: [Freeswitch-users] TTS/Cepstral choppy audio on Windows...
Message-ID: <CABWdUKDM03ACNNxMaxg1+GS72iRcj6Brwk21fvnxKNZP3kDdcQ@mail.gmail.com>

I'm building a automation control application which has an audio stream.  I
want
an IVR to allow the callee to manipulate the application while the stream
is playing.
(the audio is interfaced via soundcard/portaudio.)

I'm using the latest Build of FSClient, and interfacing to it with
EventSocket.

I'm using EventSocket to gather digits and play TTS to the caller.

The OS is Windows 7/64, and running all these apps in 32-bit.  6GB memory,
dual-core processor.
While I'm in a call, and playing TTS, CPU running at about 25% and memory
about 60%
(lots of stuff running on the box).

It does not matter if TTS is flite or cepstral, I get a bad chop at about a
1Hz or more rate.
I've tried flite and both 16 and 8-kHz cepstral voices, same issue.

Am I doing something wrong, or is this a bug?

TIA,

Gerry
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From jaganthoutam at gmail.com  Wed Apr  4 10:38:00 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Wed, 4 Apr 2012 12:08:00 +0530
Subject: [Freeswitch-users] Regarding SIP accounts creation
Message-ID: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>

Hi all,

  this is silly but i am new to Freeswitch i need to know the configuration
of sip account creation in sip_profile for local users,
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From ashishlal.work at gmail.com  Wed Apr  4 14:17:39 2012
From: ashishlal.work at gmail.com (Ashish Lal)
Date: Wed, 4 Apr 2012 15:47:39 +0530
Subject: [Freeswitch-users] help with using mod_xml_curl
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05A8C2@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE2
	8E5BBE05A512@RSV16.realitysolutions.local>	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8C2@RSV16.realitysolutions.local>
Message-ID: <4f7c1e85.0517440a.6c9c.210e@mx.google.com>

Hi,

 

I am new to freeswitch.

 

I am trying to accomplish the following - during registration our proprietary client sends a username and session id to freeswitch. 

 

Freeswitch then queries a webserver using xml_curl to authenticate the username and session id. Is this possible? Can we use mod_xml_curl to send a session id? If so how?

 

Also, I am using a simple TCP server for a dummy webserver. Do XML responses from the TCP server have to be wrapped in HTTP POST?

 

Regards,

--Ashish

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From Daniel.Knaggs at realitysolutions.co.uk  Wed Apr  4 14:25:43 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Wed, 4 Apr 2012 10:25:43 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com> 
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>

Update, it?s happened while tracing on span 1:4 so I?ve copied the in-s1c4 and out-s1c4 files ? trace is still running.  The call was inbound.

Also it appears we?re now seeing other DTMF tones in coming in as well (numbers when they haven?t been pressed on the remote end)?


Which files do you want and where you want them?


[cid:imagef33cff.PNG at b3127936.47b0e8b1]

Daniel Knaggs

Software Developer


        Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>
________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire


[cid:image2982b3.PNG at d5b7f3a2.42a6f2fb]


From: Daniel Knaggs
Sent: 04 April 2012 10:52
To: 'FreeSWITCH Users Help'
Subject: RE: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

OK, it?s now tracing on the entire span.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 03 April 2012 18:24
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

Oops, mea culpa! I meant "ftdm trace" which will record the audio on a whole span or a single channel depending on the args.
-MC
On Tue, Apr 3, 2012 at 8:52 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Audio was captured using ?record_session? where it didn?t appear in the audio by ?ear?.

?ftdm dump? displays details about the span I tell it to: -

span_id: 1
chan_id: 1
physical_span_id: 1
physical_chan_id: 1
physical_status: ok
physical_status_red: 0
physical_status_yellow: 0
physical_status_rai: 0
physical_status_blue: 0
physical_status_ais: 0
physical_status_general: 0
signaling_status: UP
type: B
state: DOWN
last_state: HANGUP_COMPLETE
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)

etc?





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From mitch.capper at gmail.com  Wed Apr  4 17:33:31 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 4 Apr 2012 06:33:31 -0700
Subject: [Freeswitch-users] TTS/Cepstral choppy audio on Windows...
In-Reply-To: <CABWdUKDM03ACNNxMaxg1+GS72iRcj6Brwk21fvnxKNZP3kDdcQ@mail.gmail.com>
References: <CABWdUKDM03ACNNxMaxg1+GS72iRcj6Brwk21fvnxKNZP3kDdcQ@mail.gmail.com>
Message-ID: <CALn233rNxS63j8UvcB3sFkP7SOO+Oxb6eK8W8=P9eSJ1Nfv00A@mail.gmail.com>

Gerry,
I forgot to ask are you using the latest binary build of FSClient or
are you building from the latest contrib repo against the latest
master of FS?  I am not sure if its related but figured might as well
make sure you are latest the latest build there.  If you want I can
send over a newer binary build.

~Mitch
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From kris at kriskinc.com  Wed Apr  4 17:54:20 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 4 Apr 2012 09:54:20 -0400
Subject: [Freeswitch-users] adding leg C for external recorder
In-Reply-To: <CAKYdwTbcfx0Up2ZKVpSKfGxx4FbwJ1JzLArwh7_zenREGRn_+g@mail.gmail.com>
References: <CAKYdwTY4xRzXMPupspGKObnYOLVEJqqDcaQpcMM904Pd-uvLBg@mail.gmail.com>
	<CALNZuwniBHYSLSNCfnz4_PrJGwRUePzTF-mAAfXgZVG_cqjHVg@mail.gmail.com>
	<CAKYdwTbMt+GD_eNvywWcKQ=z4e5Aw=FT89tyh51wzc1gmC6xgA@mail.gmail.com>
	<4F7C2E61.2050401@anew.com.ve>
	<CAKYdwTbcfx0Up2ZKVpSKfGxx4FbwJ1JzLArwh7_zenREGRn_+g@mail.gmail.com>
Message-ID: <CAKDfjgdX5cUZnOEqsbb_U=4-H5DmcGTQmoFES7Y0PA7M=cEcvw@mail.gmail.com>

I assume your FreeSWITCH system is Linux?  You could try using
iptables tee functionality:

http://www.highonphp.com/packet-cloning-with-iptables

http://www.bjou.de/blog/2008/05/howto-copyteeclone-network-traffic-using-iptables/

This will increase the network and cpu load on your FreeSWITCH machine
somewhat but if your traffic rate isn't very high it should work.
You'll probably want to use multiple rules and --dport/--sport matches
to only include SIP and RTP traffic.


On Wed, Apr 4, 2012 at 8:53 AM, Bote Man <bote_radio at botecomm.com> wrote:
> Yes, I am hoping that this Cisco supports SPAN (Switched Port Analyzer)
> because that is exactly what I need. Unfortunately, that is a job for
> someone else involved in this project due to political and business reasons,
> so I can not implement it or even check if it has that feature.
>
> The problem with eavesdrop and mod_spy is that they depend on the
> listener/recorder initiating the dialplan action. The recorder is passive so
> the stream must be sent to it as another party to the call, unless I am
> missing something here.
>
> Thanks.
>
> Bote

-- 
Kristian Kielhofner


From sharad at coraltele.com  Wed Apr  4 17:47:40 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Wed, 4 Apr 2012 19:17:40 +0530
Subject: [Freeswitch-users] Session-wise Language Setting
References: <mailman.6169.1332986617.13443.freeswitch-users@lists.freeswitch.org><4CB4E6075A9F4F9BA1C8A57FF2A06735@sharad><4F746EFF.3050305@communicatefreely.net><00C2DD790B254A1BACD50CAA66B123B9@sharad><1FFF97C269757C458224B7C895F35F15083A49@cantor.std.visionutv.se><12BA6FAF92674DEE9650CF4829B95DF0@sharad>	<CAKzWOxWus_Ryo97ARrJYbFjaJrmPEUxDF169iqzLKbw7AaW+DQ@mail.gmail.com><13D710C546A14C09801E1CC46A394BB4@sharad><4F79CA93.3030205@communicatefreely.net>
	<5A93FC0084F946DCB75E8F139E53DD40@sharad>
Message-ID: <7D6D8BF360064326B4011EE4EA95656A@sharad>

Dear Mr. Tim,

Your suggested way has worked. Thanks for your kind help.

Best Regards
Sharad


----- Original Message ----- 
From: "Sharad Garg" <sharad at coraltele.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, April 03, 2012 12:44 PM
Subject: Re: [Freeswitch-users] Session-wise Language Setting


> Hello Mr. Tim
>
> Thanks a ton for your valuable suggestion & time.
>
> I think I found the solution of my query. I will definately try this & let 
> you know the feedback.
>
> Thanks a lot once again.
>
> Take care n by.
>
> Sharad
>
>
> ----- Original Message ----- 
> From: "Tim St. Pierre" <fs-list at communicatefreely.net>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, April 02, 2012 9:19 PM
> Subject: Re: [Freeswitch-users] Session-wise Language Setting
>
>
>> Hi Sharad,
>>
>> When you specify a sound file to play like that, it uses a prefix defined 
>> (usually) in vars.xml, and it doesn't expand to the language.  The 
>> assumption by the designers I assume was that you would change the prefix 
>> to your language.
>>
>> For multi-language systems, you should always use phrase macros.
>>
>> On our system, we made several custom phrase macros that are very simple, 
>> but will select the correct audio file based on the language.
>>
>> Any time you would normally specify an audio file, you can instead use 
>> phrase:my_macro:data
>>
>> Then, create a phrase macro for each language that will match your data 
>> and play the appropriate audio file.
>>
>> It could be something as simple as this:
>> English:
>>
>> <macro name="ivr-file">
>>      <input pattern="^(\w+)$">
>>        <match>
>>            <action function="play-file" 
>> data="$${media_dir}/en/us/callie/ivr/$1wav"/>
>>        </match>
>>      </input>
>> </macro
>>
>> And then in your Russian phrase macros:
>>
>> <macro name="ivr-file">
>>      <input pattern="^(\w+)$">
>>        <match>
>>            <action function="play-file" 
>> data="$${media_dir}/ru/RU/elena/ivr/$1wav"/>
>>        </match>
>>      </input>
>> </macro
>>
>> Be creative here.  Phrase macros are your friend.  I would limit playing 
>> audio files to things like sound effects, or other non-verbal recordings.
>>
>>
>> Sharad Garg wrote:
>>>
>>> Hello Mr. MC
>>>
>>> Thanks for the link.....seems working.
>>>
>>> But observed another point i.e. it works only for the speech phrase 
>>> macros.
>>>
>>> When I write a simple xml dialplan like this -
>>>
>>>     <extension name="lang_test">
>>>          <condition field="destination_number" expression="^(3333)$">
>>>               <action application="set" data="bypass_media=true"/>
>>>               <action application="answer"/>
>>>               <action application="sleep" data="1000"/>
>>>               <action application="set" data="default_language=ru"/>
>>>               <action application="playback" data="ivr/test.wav"/>
>>>          </condition>
>>>     </extension>
>>>
>>> This dialplan still plays the test.wav which is in 
>>> <base_dir>/sounds/en/us/callie/ivr/8000/test.wav. While it should play 
>>> the test.wav from <base_dir>/sounds/ru/RU/elena/ivr/8000/test.wav. & at 
>>> the same time, speech phrase macro use the 'ru' language.
>>>
>>> So when I did my googling, I found that the language settings is only 
>>> for speech phrase macros. It is not implemented in Freeswitch to use the 
>>> desired languages at all places.
>>>
>>> So need to know is there any latest development on this ?
>>>
>>> Thanks in advance.
>>>
>>> Regards
>>>
>>> Sharad
>>>
>>>
>>>     ----- Original Message -----
>>>     *From:* Michael Collins <mailto:msc at freeswitch.org>
>>>     *To:* FreeSWITCH Users Help
>>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>>     *Sent:* Saturday, March 31, 2012 1:15 AM
>>>     *Subject:* Re: [Freeswitch-users] Session-wise Language Setting
>>>
>>>
>>>
>>>     On Fri, Mar 30, 2012 at 5:13 AM, Sharad Garg <sharad at coraltele.com
>>>     <mailto:sharad at coraltele.com>> wrote:
>>>
>>>         Thanks Peter...yes it works.
>>>
>>>         Just a last query - Is there any API for setting the same so
>>>         that it can be
>>>         used from Javascript.
>>>
>>>         Best Regards
>>>         Sharad
>>>
>>>
>>>     http://wiki.freeswitch.org/wiki/Mod_commands#uuid_setvar
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>
>
>
> 



From gerry at pstn2.net  Wed Apr  4 18:16:52 2012
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 4 Apr 2012 10:16:52 -0400
Subject: [Freeswitch-users] TTS/Cepstral choppy audio on Windows...
In-Reply-To: <CALn233rNxS63j8UvcB3sFkP7SOO+Oxb6eK8W8=P9eSJ1Nfv00A@mail.gmail.com>
References: <CABWdUKDM03ACNNxMaxg1+GS72iRcj6Brwk21fvnxKNZP3kDdcQ@mail.gmail.com>
	<CALn233rNxS63j8UvcB3sFkP7SOO+Oxb6eK8W8=P9eSJ1Nfv00A@mail.gmail.com>
Message-ID: <CABWdUKDf+sr1U6SrzN2Bk73zocmDtiMt+zogkOkvRapj-RZaCw@mail.gmail.com>

Hi Mitch,

I installed http://files.freeswitch.org/windows/installer/x86/FSClient.zip

I'm not sure how new that is.

I'll want to d/l your source later... right now just trying to prove out
some concepts.

I tried to build the full FS using the _2010 sln last night, but it
generated a lot of errors,
even after following the instructions in the wiki.  I'll try and figure
that out later.

Please, send the latest binary and I'll try that.

I was able to build the mod_flite and mod_cepstral, so I know they are the
latest.

-Gerry

On Wed, Apr 4, 2012 at 9:33 AM, Mitch Capper <mitch.capper at gmail.com> wrote:

> Gerry,
> I forgot to ask are you using the latest binary build of FSClient or
> are you building from the latest contrib repo against the latest
> master of FS?  I am not sure if its related but figured might as well
> make sure you are latest the latest build there.  If you want I can
> send over a newer binary build.
>
> ~Mitch
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From freeswitch-list at puzzled.xs4all.nl  Wed Apr  4 18:57:55 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 04 Apr 2012 16:57:55 +0200
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
Message-ID: <4F7C6173.4040000@puzzled.xs4all.nl>

On 04/04/2012 12:25 PM, Daniel Knaggs wrote:

As has been hinted before, would you mind no longer sending that over 
the top email signature? Every time I see it I ignore or even delete the 
email. And I doubt I'm the only one.

If it's your company's policy to insist on sending such a signature (so 
you can't disable it) then just get a Gmail account and subscribe to the 
list from your Gmail account. Easier on the eyes of the list subscribers 
and you might even get an answer to your question :-)

Thanks,
Patrick


From fernando.berretta at gmail.com  Wed Apr  4 19:05:18 2012
From: fernando.berretta at gmail.com (Fernando Berretta)
Date: Wed, 04 Apr 2012 12:05:18 -0300
Subject: [Freeswitch-users] Busqueda de experto en FreeSwitch en Cordoba
	Argentina
In-Reply-To: <1325787602.37602.YahooMailClassic@web2804.biz.mail.ne1.yahoo.com>
References: <1325787602.37602.YahooMailClassic@web2804.biz.mail.ne1.yahoo.com>
Message-ID: <4F7C632E.6010203@gmail.com>

Hola,

Hay en el grupo alg?n experto de FreeSwitch en C?rdoba Argentina ?  de 
ser as? por favor contactarse en privado a rrhh at voipexperts.com.ar

Saludos,
Fernando


From babak.freeswitch at gmail.com  Wed Apr  4 19:29:01 2012
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 4 Apr 2012 19:59:01 +0430
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
	cmd="exec" ...
Message-ID: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>

Hi
I'm trying to setup a centralized dialplan for  multiple instances of
freeswitch. I tried
<X-PRE-PROCESS cmd="exec" data="wget -qO -
http://www.example.com/dialplan.xml" />
but it is not implemented on windows yet. is there any similar alternative?
thanx
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From bdfoster at endigotech.com  Wed Apr  4 19:29:18 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 4 Apr 2012 11:29:18 -0400
Subject: [Freeswitch-users] double nat trouble
In-Reply-To: <00f501cd1239$ccfd7190$66f854b0$@launch3.net>
References: <00f501cd1239$ccfd7190$66f854b0$@launch3.net>
Message-ID: <CAKV-jFw8=sBEdYXt9_LRn3MvsxUNMkakyu2JnzQaOP+7_jTmcg@mail.gmail.com>

Please attach a .pcap from the FS server with a failed call.

-BDF
On Apr 4, 2012 4:06 AM, "Brett Wilson" <brett at launch3.net> wrote:

> Hey guys,****
>
> We have just temporarily moved our office, so our phones are no longer
> just on the office LAN. They are on a remote lan, both sides behind NAT and
> I?m trying to get them working right but I am having trouble. I have things
> working to the point where I can call to the POTS or to an extension on the
> local network back home just fine. The problem is that the remote phones do
> not receive sip messages behind the nat. It seems to me that freeswitch is
> sending the messages to the sip port the phones are configured with, and
> not the NAT port that can be seen when viewing the registrations status.
> The nat ports are very high numbers. I suspect if FS was to send messages
> to those ports, it would get through the foreign nat and reach the phone.*
> ***
>
> ** **
>
> On the FS side, I am using pfsense as the firewall, and have port forwards
> set up to the fs box. I am using a dynamic ip with dyn-dns-like service and
> am using that domain name in the fs internal/external profiles for the
> ext-sip-ip and ext-rtp-ip variables. I have upnp turned off but it does not
> make any difference anyway.****
>
> ** **
>
> Strangely, I was actually able to place a call from one remote phone to
> another once earlier, but it must have been a fluke because I haven?t had
> it working after that.****
>
> ** **
>
> What am I missing here?****
>
> ** **
>
> *******************************************
> *Brett Wilson*
> *IT Department*
> *Launch 3 Ventures, LLC*
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866
> *Email:* Brett.Wilson at launch3.net
> *AOL IM:* Brett.Wilson at launch3.net
> www.Launch3.net <http://www.launch3.net/>
> *www.Launch3telecom.com <http://www.launch3telecom.com/>*
> *******************************************
> [image: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>[image:
> Description: Description: GPlus-Logo]<https://plus.google.com/102713212642017290959>
> ****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Wed Apr  4 19:31:29 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 4 Apr 2012 11:31:29 -0400
Subject: [Freeswitch-users] Regarding SIP accounts creation
In-Reply-To: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
References: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
Message-ID: <CAKV-jFzeRWSj49_2XEpFNObfPfuW2gjXaOaM_K8TaF5K1LAZKQ@mail.gmail.com>

Please help us help you by taking a look on the wiki.

-BDF
On Apr 4, 2012 9:21 AM, "Jagadish Thoutam" <jaganthoutam at gmail.com> wrote:

> Hi all,
>
>   this is silly but i am new to Freeswitch i need to know the
> configuration of sip account creation in sip_profile for local users,
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Wed Apr  4 19:37:48 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 4 Apr 2012 18:37:48 +0300
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
Message-ID: <CALNZuwmRzsD46mYgPLMK6ByDGENCMM2od8WNvRpBNL339kKUaw@mail.gmail.com>

You can probably get a drop-in wget replacement for windows.
(Or curl with appropriate params?)

-Avi

On Wed, Apr 4, 2012 at 6:29 PM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> Hi
> I'm trying to setup a centralized dialplan for  multiple instances of
> freeswitch. I tried
> <X-PRE-PROCESS cmd="exec" data="wget -qO -
> http://www.example.com/dialplan.xml" />
> but it is not implemented on windows yet. is there any similar alternative?
> thanx
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Wed Apr  4 19:38:31 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 4 Apr 2012 11:38:31 -0400
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
Message-ID: <CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>

Wget is a Linux system command. Its not going to ever work on windows.

-BDF
On Apr 4, 2012 11:30 AM, "babak yakhchali" <babak.freeswitch at gmail.com>
wrote:

> Hi
> I'm trying to setup a centralized dialplan for  multiple instances of
> freeswitch. I tried
> <X-PRE-PROCESS cmd="exec" data="wget -qO -
> http://www.example.com/dialplan.xml" />
> but it is not implemented on windows yet. is there any similar alternative?
> thanx
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vetali100 at gmail.com  Wed Apr  4 19:39:10 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 4 Apr 2012 08:39:10 -0700
Subject: [Freeswitch-users] double nat trouble
In-Reply-To: <00f501cd1239$ccfd7190$66f854b0$@launch3.net>
References: <00f501cd1239$ccfd7190$66f854b0$@launch3.net>
Message-ID: <CACc3D0WPO+tFuywozCA0=SqSu6T0uGMptX3Z2wdHcYy=aOAHiw@mail.gmail.com>

Try to set NDLB-force-rport = true in your internal profile and see if that
helps.

NDLB-force-rport

This will force FreeSWITCH to send SIP responses to the network port from
which they were received. *Use at your own risk!* For more information see NAT
Traversal <http://wiki.freeswitch.org/wiki/NAT_Traversal>.

<param name="NDLB-force-rport" value="true|safe"/>


http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-force-rport

Regards,
Vitalie

2012/4/4 Brett Wilson <brett at launch3.net>

> Hey guys,****
>
> We have just temporarily moved our office, so our phones are no longer
> just on the office LAN. They are on a remote lan, both sides behind NAT and
> I?m trying to get them working right but I am having trouble. I have things
> working to the point where I can call to the POTS or to an extension on the
> local network back home just fine. The problem is that the remote phones do
> not receive sip messages behind the nat. It seems to me that freeswitch is
> sending the messages to the sip port the phones are configured with, and
> not the NAT port that can be seen when viewing the registrations status.
> The nat ports are very high numbers. I suspect if FS was to send messages
> to those ports, it would get through the foreign nat and reach the phone.*
> ***
>
> ** **
>
> On the FS side, I am using pfsense as the firewall, and have port forwards
> set up to the fs box. I am using a dynamic ip with dyn-dns-like service and
> am using that domain name in the fs internal/external profiles for the
> ext-sip-ip and ext-rtp-ip variables. I have upnp turned off but it does not
> make any difference anyway.****
>
> ** **
>
> Strangely, I was actually able to place a call from one remote phone to
> another once earlier, but it must have been a fluke because I haven?t had
> it working after that.****
>
> ** **
>
> What am I missing here?****
>
> ** **
>
> *******************************************
> *Brett Wilson*
> *IT Department*
> *Launch 3 Ventures, LLC*
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866
> *Email:* Brett.Wilson at launch3.net
> *AOL IM:* Brett.Wilson at launch3.net
> www.Launch3.net <http://www.launch3.net/>
> *www.Launch3telecom.com <http://www.launch3telecom.com/>*
> *******************************************
> [image: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>[image:
> Description: Description: GPlus-Logo]<https://plus.google.com/102713212642017290959>
> ****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From cesar.bermudez at gmail.com  Wed Apr  4 19:41:05 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Wed, 4 Apr 2012 12:41:05 -0300
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
Message-ID: <CADrcvV8uEsiXcBgwGKFBC8k+Ojcqm507nkoAFWesHk2FGMCPGA@mail.gmail.com>

You have wget for windows ...
i dont say that works, but you have it.

On Wed, Apr 4, 2012 at 12:38 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Wget is a Linux system command. Its not going to ever work on windows.
>
> -BDF
> On Apr 4, 2012 11:30 AM, "babak yakhchali" <babak.freeswitch at gmail.com>
> wrote:
>
>> Hi
>> I'm trying to setup a centralized dialplan for  multiple instances of
>> freeswitch. I tried
>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>> http://www.example.com/dialplan.xml" />
>> but it is not implemented on windows yet. is there any similar
>> alternative?
>> thanx
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jkomar at jbox.ca  Wed Apr  4 19:44:09 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Wed, 4 Apr 2012 09:44:09 -0600
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
Message-ID: <CA+zebiYOhOhgcjuQst39h8qbMo_fniZzw5P2Jnt1N86vqHtndg@mail.gmail.com>

wget for windows:

http://gnuwin32.sourceforge.net/packages/wget.htm

Jason

On Wed, Apr 4, 2012 at 9:38 AM, Brian Foster <bdfoster at endigotech.com> wrote:
> Wget is a Linux system command. Its not going to ever work on windows.
>
> -BDF
>
> On Apr 4, 2012 11:30 AM, "babak yakhchali" <babak.freeswitch at gmail.com>
> wrote:
>>
>> Hi
>> I'm trying to setup a centralized dialplan for ?multiple instances of
>> freeswitch. I tried
>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>> http://www.example.com/dialplan.xml" />
>> but it is not implemented on windows yet. is there any similar
>> alternative?
>> thanx
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bdfoster at endigotech.com  Wed Apr  4 19:47:15 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 4 Apr 2012 11:47:15 -0400
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CA+zebiYOhOhgcjuQst39h8qbMo_fniZzw5P2Jnt1N86vqHtndg@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
	<CA+zebiYOhOhgcjuQst39h8qbMo_fniZzw5P2Jnt1N86vqHtndg@mail.gmail.com>
Message-ID: <CAKV-jFyHHXuX_eEzpx4EXx3Gt1wY25qNvpskj8LqhQKaf2mkfA@mail.gmail.com>

...and I just proved to the world I don't know a damn thing about Windows...

-BDF
On Apr 4, 2012 11:45 AM, "Komar, Jason" <jkomar at jbox.ca> wrote:

> wget for windows:
>
> http://gnuwin32.sourceforge.net/packages/wget.htm
>
> Jason
>
> On Wed, Apr 4, 2012 at 9:38 AM, Brian Foster <bdfoster at endigotech.com>
> wrote:
> > Wget is a Linux system command. Its not going to ever work on windows.
> >
> > -BDF
> >
> > On Apr 4, 2012 11:30 AM, "babak yakhchali" <babak.freeswitch at gmail.com>
> > wrote:
> >>
> >> Hi
> >> I'm trying to setup a centralized dialplan for  multiple instances of
> >> freeswitch. I tried
> >> <X-PRE-PROCESS cmd="exec" data="wget -qO -
> >> http://www.example.com/dialplan.xml" />
> >> but it is not implemented on windows yet. is there any similar
> >> alternative?
> >> thanx
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Hector.Geraldino at ipsoft.com  Wed Apr  4 20:29:22 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Wed, 4 Apr 2012 12:29:22 -0400
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAKV-jFyHHXuX_eEzpx4EXx3Gt1wY25qNvpskj8LqhQKaf2mkfA@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAKV-jFwjM_Mm-WZ9d-xNrLSNc1AN+0_9Ok=AgTMhmZSNzA+eBQ@mail.gmail.com>
	<CA+zebiYOhOhgcjuQst39h8qbMo_fniZzw5P2Jnt1N86vqHtndg@mail.gmail.com>
	<CAKV-jFyHHXuX_eEzpx4EXx3Gt1wY25qNvpskj8LqhQKaf2mkfA@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022D77D6DF@NY1-EXMB-01.ip-soft.net>

Install cygwin, it contains wget among many other unix system tools.

http://www.cygwin.com/


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Wednesday, April 04, 2012 11:47 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] windows alternative for <X-PRE-PROCESS cmd="exec" ...


...and I just proved to the world I don't know a damn thing about Windows...

-BDF
On Apr 4, 2012 11:45 AM, "Komar, Jason" <jkomar at jbox.ca<mailto:jkomar at jbox.ca>> wrote:
wget for windows:

http://gnuwin32.sourceforge.net/packages/wget.htm

Jason

On Wed, Apr 4, 2012 at 9:38 AM, Brian Foster <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:
> Wget is a Linux system command. Its not going to ever work on windows.
>
> -BDF
>
> On Apr 4, 2012 11:30 AM, "babak yakhchali" <babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>>
> wrote:
>>
>> Hi
>> I'm trying to setup a centralized dialplan for  multiple instances of
>> freeswitch. I tried
>> <X-PRE-PROCESS cmd="exec" data="wget -qO -
>> http://www.example.com/dialplan.xml" />
>> but it is not implemented on windows yet. is there any similar
>> alternative?
>> thanx
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From gerry at pstn2.net  Wed Apr  4 21:15:05 2012
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 4 Apr 2012 13:15:05 -0400
Subject: [Freeswitch-users] TTS/Cepstral choppy audio on Windows...
In-Reply-To: <CABWdUKDf+sr1U6SrzN2Bk73zocmDtiMt+zogkOkvRapj-RZaCw@mail.gmail.com>
References: <CABWdUKDM03ACNNxMaxg1+GS72iRcj6Brwk21fvnxKNZP3kDdcQ@mail.gmail.com>
	<CALn233rNxS63j8UvcB3sFkP7SOO+Oxb6eK8W8=P9eSJ1Nfv00A@mail.gmail.com>
	<CABWdUKDf+sr1U6SrzN2Bk73zocmDtiMt+zogkOkvRapj-RZaCw@mail.gmail.com>
Message-ID: <CABWdUKD2+zacL4xJgaS9c_C6fKYtQEHG=bON9ov-WJKde+1vEA@mail.gmail.com>

So, here's what's in my log:

Why all the " switch_core_io.c:883 Deactivating write resampler " messages?
Its this what is causing all the stutter?

2012-04-04 13:12:05.385377 [DEBUG] switch_ivr_play_say.c:2392 OPEN TTS
cepstral
2012-04-04 13:12:05.385377 [DEBUG] switch_ivr_play_say.c:2401 Raw Codec
Activated
2012-04-04 13:12:05.385377 [DEBUG] switch_ivr_play_say.c:2092 Speaking
text: you pressed the number 6
2012-04-04 13:12:05.405378 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.425380 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.445381 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.465382 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.485383 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.505384 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.525385 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.545386 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.565388 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.585389 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.625391 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.745398 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.785400 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.885406 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.925408 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:05.965410 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.025414 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.165422 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.225425 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.305430 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.445438 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.505441 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.585446 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.605447 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.725454 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.865462 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.885463 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:06.925465 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.005470 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.065473 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.145478 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.225483 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.285486 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.425494 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.485497 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.565502 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.705510 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.765513 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.845518 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.905521 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:07.985526 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.125534 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.265542 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.405550 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.465553 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.525557 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.685566 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.825574 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.845575 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:08.965582 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.025585 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.045587 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.085589 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.185595 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.245598 [NOTICE] switch_core_io.c:883 Deactivating write
resampler
2012-04-04 13:12:09.265599 [DEBUG] switch_ivr_play_say.c:2284 done speaking
text
2012-04-04 13:12:09.645621 [DEBUG] switch_core_media_bug.c:508 Removing BUG
from portaudio/6036246907


On Wed, Apr 4, 2012 at 10:16 AM, Gerry Hull <gerry at pstn2.net> wrote:

> Hi Mitch,
>
> I installed http://files.freeswitch.org/windows/installer/x86/FSClient.zip
>
> I'm not sure how new that is.
>
> I'll want to d/l your source later... right now just trying to prove out
> some concepts.
>
> I tried to build the full FS using the _2010 sln last night, but it
> generated a lot of errors,
> even after following the instructions in the wiki.  I'll try and figure
> that out later.
>
> Please, send the latest binary and I'll try that.
>
> I was able to build the mod_flite and mod_cepstral, so I know they are the
> latest.
>
> -Gerry
>
> On Wed, Apr 4, 2012 at 9:33 AM, Mitch Capper <mitch.capper at gmail.com>wrote:
>
>> Gerry,
>> I forgot to ask are you using the latest binary build of FSClient or
>> are you building from the latest contrib repo against the latest
>> master of FS?  I am not sure if its related but figured might as well
>> make sure you are latest the latest build there.  If you want I can
>> send over a newer binary build.
>>
>> ~Mitch
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From nasida at live.ru  Wed Apr  4 21:20:51 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 4 Apr 2012 21:20:51 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
	sofia_reg_internal.db
Message-ID: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>


Hello guys. 
I have issue with incorrect reg rows in sqlite database sofia_reg_internal.db. it looks like that one registrations are crossed with others. so sometimes i have 2 inbound call for 2 different users instead of one. What it can be ? probably it is some bug of new FS version ? Can you give me some advise ? 
sqlite> select * from sip_registrations where sip_user='user2';
3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||user1 <sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2 <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101

Why I have first row here?  When I call for user2 I have two call (for user1 an user2) instead of one (for user2).I have separate row for user1...
Thanks. 		 	   		  
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From msc at freeswitch.org  Wed Apr  4 21:34:49 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 Apr 2012 10:34:49 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxXVEzZcR80qvW1aKOzM-3EY4rBHH8GTLxAorGxFCmmeVQ@mail.gmail.com>

On Wed, Apr 4, 2012 at 3:25 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

>   Update, it?s happened while tracing on span 1:4 so I?ve copied the
> in-s1c4 and out-s1c4 files ? trace is still running.  The call was inbound.
> ****
>
> ** **
>
> Also it appears we?re now seeing other DTMF tones in coming in as well
> (numbers when they haven?t been pressed on the remote end)?****
>
> ** **
>
> ** **
>
> Which files do you want and where you want them?****
>
> **
>

I recommend that you open a ticket w/ Sangoma and ask them to analyze the
traces. The critical thing is to find out if these are false positives,
phantoms, or stuff actually coming down the line from the carrier. I think
Sangoma is in the best position to help.
-MC

P.S. - Regarding the footer - yes, it is annoying, but if you can't turn it
off then you can't turn it off. I recommend doing what Patrick suggests and
getting a gmail account so that you can be free of the footer spam. If you
*really* want to help us out, I'd say investigate the Mailman project and
see if it has a feature that can auto-strip the junk off of the end of an
email. :)
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From mario_fs at mgtech.com  Wed Apr  4 21:51:11 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 10:51:11 -0700
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
Message-ID: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>

Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done?
Mario G

        <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
	<action application="set" data="ringback=${us-ring}"/>
	<action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>

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From nasida at live.ru  Wed Apr  4 21:55:56 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 4 Apr 2012 21:55:56 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
 sofia_reg_internal.db
In-Reply-To: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
Message-ID: <SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>


Update.
after ./fs_cli -x "sofia profile internal flush_inbound_reg"  I have
sqlite> select * from sip_registrations where sip_user='
user2 ';|
user2 ||||||||||||||||||||
Why?? I must have clear rows!!
Next, I will have 2 rows again but with user3 instead of user1...
Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 4 Apr 2012 21:20:51 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database	sofia_reg_internal.db







Hello guys. 
I have issue with incorrect reg rows in sqlite database sofia_reg_internal.db. it looks like that one registrations are crossed with others. so sometimes i have 2 inbound call for 2 different users instead of one. What it can be ? probably it is some bug of new FS version ? Can you give me some advise ? 
sqlite> select * from sip_registrations where sip_user='user2';
3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||  user1 <sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2 <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101

Why I have first row here?  When I call for user2 I have two call (for user1 an user2) instead of one (for user2).I have separate row for user1...
Thanks. 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Apr  4 21:56:32 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 4 Apr 2012 12:56:32 -0500
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
	sofia_reg_internal.db
In-Reply-To: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
Message-ID: <CAKbxfG_V0-Gj0ede_qaO4Xd4qH4cMH-+LAigMtOrdpYQ68QPtA@mail.gmail.com>

review your configs on the device and check for inconsistencies.
enter sofia global siptrace on from cli and look at the REGISTER packet.

On Wed, Apr 4, 2012 at 12:20 PM, Yuriy Nasida <nasida at live.ru> wrote:
> Hello guys.
>
> I have issue with incorrect reg rows in sqlite database
> sofia_reg_internal.db. it looks like that one registrations are crossed with
> others. so sometimes i have 2 inbound call for 2 different users instead of
> one. What it can be ? probably it is some bug of new FS version ? Can you
> give me some advise ?
>
> sqlite> select * from sip_registrations where sip_user='user2';
>
> 3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||user1
> <sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>
> 3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2
> <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>
>
> Why I have first row here? ?When I call for?user2 I have two call (for?user1
> an?user2) instead of one (for?user2).
> I have separate row for user1...
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From gabe at gundy.org  Wed Apr  4 22:10:40 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 12:10:40 -0600
Subject: [Freeswitch-users] Regarding SIP accounts creation
In-Reply-To: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
References: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
Message-ID: <CAMwZ-tJE-u9Phwj1PUhMeLTiv5AUDf1D1COro+z_85HKgnG6Kw@mail.gmail.com>

On Wed, Apr 4, 2012 at 12:38 AM, Jagadish Thoutam
<jaganthoutam at gmail.com> wrote:
> this is silly but i am new to Freeswitch i need to know the configuration of
> sip account creation in sip_profile for local users,

I'm not sure what you're looking for, but if you're looking to enable
users to register their phones, you actually want the Directory:
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide

In the source code, have a look at this dir for some pointers:
conf/vanilla/directory/


Good luck,
Gabe


From gabe at gundy.org  Wed Apr  4 22:11:36 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 12:11:36 -0600
Subject: [Freeswitch-users] Regarding SIP accounts creation
In-Reply-To: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
References: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
Message-ID: <CAMwZ-tJHM484L03MeLvP7g=3TZ=bnJEjtj=QykKbJAA95Z_N+A@mail.gmail.com>

On Wed, Apr 4, 2012 at 12:38 AM, Jagadish Thoutam
<jaganthoutam at gmail.com> wrote:
> this is silly but i am new to Freeswitc

Oh, and welcome to FreeSWITCH!!!


Best,
Gabe


From jaganthoutam at gmail.com  Wed Apr  4 22:16:01 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Wed, 4 Apr 2012 23:46:01 +0530
Subject: [Freeswitch-users] Regarding SIP accounts creation
In-Reply-To: <CAMwZ-tJHM484L03MeLvP7g=3TZ=bnJEjtj=QykKbJAA95Z_N+A@mail.gmail.com>
References: <CAHqn9HFs5KCQX12VVmrCM9OyfYgA_mQguCYTj+NvFCv5=LqPgw@mail.gmail.com>
	<CAMwZ-tJHM484L03MeLvP7g=3TZ=bnJEjtj=QykKbJAA95Z_N+A@mail.gmail.com>
Message-ID: <CAHqn9HH-6kF0=fT3fe4-fLV-PqXrLXR0FW1JKEDt57euopBvNA@mail.gmail.com>

thank you, i got it now going good its very interesting

On 4 April 2012 23:41, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Apr 4, 2012 at 12:38 AM, Jagadish Thoutam
> <jaganthoutam at gmail.com> wrote:
> > this is silly but i am new to Freeswitc
>
> Oh, and welcome to FreeSWITCH!!!
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From nasida at live.ru  Wed Apr  4 22:26:46 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 4 Apr 2012 22:26:46 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
 sofia_reg_internal.db
In-Reply-To: <SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>,
	<SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>
Message-ID: <SNT127-W65713C002F6E87F9869B13D5320@phx.gbl>


Anthony, thank you for your answer but I think it is not issue with settings of sip devices. Incorrect additional row may be any of my customers. I mean that the additional row do change after   ./fs_cli -x "sofia profile internal flush_inbound_reg"
Also I have this problem after moving to : FreeSWITCH Version 1.0.head (git-2ec02e5 2012-03-01 13-22-52 -0600)  + Centos 6.
I really don't understand why i have it just after cleaning of all rows...
sqlite> select * from sip_registrations where sip_user=' user2 ';| user2 ||||||||||||||||||||
Please advice.
Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 4 Apr 2012 21:55:56 +0400
Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database sofia_reg_internal.db







Update.
after ./fs_cli -x "sofia profile internal flush_inbound_reg"  I have
sqlite> select * from sip_registrations where sip_user='
user2 ';|
user2 ||||||||||||||||||||
Why?? I must have clear rows!!
Next, I will have 2 rows again but with user3 instead of user1...
Thanks.

From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 4 Apr 2012 21:20:51 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database	sofia_reg_internal.db







Hello guys. 
I have issue with incorrect reg rows in sqlite database sofia_reg_internal.db. it looks like that one registrations are crossed with others. so sometimes i have 2 inbound call for 2 different users instead of one. What it can be ? probably it is some bug of new FS version ? Can you give me some advise ? 
sqlite> select * from sip_registrations where sip_user='user2';
3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||  user1 <sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2 <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101

Why I have first row here?  When I call for user2 I have two call (for user1 an user2) instead of one (for user2).I have separate row for user1...
Thanks. 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Wed Apr  4 22:51:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 4 Apr 2012 13:51:02 -0500
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
	sofia_reg_internal.db
In-Reply-To: <SNT127-W65713C002F6E87F9869B13D5320@phx.gbl>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
	<SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>
	<SNT127-W65713C002F6E87F9869B13D5320@phx.gbl>
Message-ID: <CAKbxfG8bYVwe+LFkDoa6tWA2KoF3V8vy4b8iE6m6mywWvdGDPw@mail.gmail.com>

I already advised.

You did the first one and checked your settings.
Next capture sip traffic

sofia global siptrace on

and look at all the registers for anything anomalous

On Wed, Apr 4, 2012 at 1:26 PM, Yuriy Nasida <nasida at live.ru> wrote:
> Anthony, thank you for your answer but I think it is not issue with settings
> of sip devices. Incorrect additional row may be any of my customers. I mean
> that the additional row do change after ? ./fs_cli -x "sofia profile
> internal flush_inbound_reg"
>
> Also I have this problem after moving to : FreeSWITCH Version 1.0.head
> (git-2ec02e5 2012-03-01 13-22-52 -0600) ?+ Centos 6.
>
> I really don't understand why i have it just?after cleaning of all rows...
>
> sqlite> select * from sip_registrations where sip_user=' user2 ';
> | user2 ||||||||||||||||||||
>
> Please advice.
>
> Thanks.
>
> ________________________________
> From: nasida at live.ru
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 4 Apr 2012 21:55:56 +0400
> Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database
> sofia_reg_internal.db
>
>
> Update.
>
> after?./fs_cli -x "sofia profile internal flush_inbound_reg" ?I have
>
> sqlite> select * from sip_registrations where sip_user=' user2?';
> | user2?||||||||||||||||||||
>
> Why?? I must have clear rows!!
>
> Next, I will have 2 rows again but with?user3 instead of?user1...
>
> Thanks.
>
> ________________________________
> From: nasida at live.ru
> To: freeswitch-users at lists.freeswitch.org
> Date: Wed, 4 Apr 2012 21:20:51 +0400
> Subject: [Freeswitch-users] incorrect reg rows in sqlite database
> sofia_reg_internal.db
>
> Hello guys.
>
> I have issue with incorrect reg rows in sqlite database
> sofia_reg_internal.db. it looks like that one registrations are crossed with
> others. so sometimes i have 2 inbound call for 2 different users instead of
> one. What it can be ? probably it is some bug of new FS version ? Can you
> give me some advise ?
>
> sqlite> select * from sip_registrations where sip_user='user2';
>
> 3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||??user1?<sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>
> 3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2
> <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>
>
> Why I have first row here? ?When I call for?user2 I have two call (for?user1
> an?user2) instead of one (for?user2).
> I have separate row for user1...
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From admharris at gmx.com  Thu Apr  5 00:11:52 2012
From: admharris at gmx.com (adam harris)
Date: Wed, 04 Apr 2012 22:11:52 +0200
Subject: [Freeswitch-users] DTMF event only captured on playback
Message-ID: <20120404201152.27680@gmx.com>


I have set eventconsumer to all and I am able to capture events.

However DTMF events can only be received while playback is going on from a session execute playback. Once the file has stop playing, DTMF events are no longer caught. It seems somthing is set during playback.

What is needed in order to have DTMFS caputured as a event to a con:pop function even whilst no playback is running?


From mario_fs at mgtech.com  Thu Apr  5 00:21:37 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 13:21:37 -0700
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
In-Reply-To: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
Message-ID: <ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>

I tried putting the ringback into a bridge command - no effect, I also tried adding bridge_early_media to the bridge - no effect. The problem is when the call  is picked during the the second bridge which is used to create a different ring on the extensions (there is a later bridge to answer) the other phones in the group continue to ring. I am trying to solve this which for some reason occurs more after I updated to the recent git. I am hoping if I can combine into a single bridge command the problem will be solved. Any suggestion greatly appreciated.

BTW, the command below does nothing, nothing in the debug, it shows up but nothing happens. I tried using ! , | but no luck.

   <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>


On Apr 4, 2012, at 10:51 AM, Mario G wrote:

> Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done?
> Mario G
> 
>         <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
> 	<action application="set" data="ringback=${us-ring}"/>
> 	<action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>
> 
> _________________________________________________________________________

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From gabe at gundy.org  Thu Apr  5 01:08:43 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 15:08:43 -0600
Subject: [Freeswitch-users] Handing calls coming from Kamailio
In-Reply-To: <CAG2QdgbgpQ4mnnaPfGk1zvzAjVWyP7G5_-B5TpmXd68Cy588-A@mail.gmail.com>
References: <CAG2QdgbgpQ4mnnaPfGk1zvzAjVWyP7G5_-B5TpmXd68Cy588-A@mail.gmail.com>
Message-ID: <CAMwZ-tLqa_HxqqERd=EQ9jsbJVPs_p-SnNfLfUO7Hb67MYSH1A@mail.gmail.com>

On Wed, Apr 4, 2012 at 5:36 AM, Ali Jawad <ali.jawad at splendor.net> wrote:
> Any hints on what I have missed please ?

We're going to need more info.  Logs from FreeSWITCH would be a good
place to start.  There are nearly an infinite number of options when
configuring FreeSWITCH, SIP proxies and gateways. We can help unless
you give us more of an idea about what you're seeing.

Also, a look at your dialplan and possibly some SIP traces would go a
long way toward helping.


Best,
Gabe


From msc at freeswitch.org  Thu Apr  5 01:33:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 Apr 2012 14:33:10 -0700
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
In-Reply-To: <ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
Message-ID: <CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>

What's the plain language description of what you're trying to do? I think
we're all a bit confused about exactly which problem you're attempting to
solve. A plain language description (without any telephonese) will
definitely get us all on the same page. :)

Thanks,
MC

On Wed, Apr 4, 2012 at 1:21 PM, Mario G <mario_fs at mgtech.com> wrote:

> I tried putting the ringback into a bridge command - no effect, I also
> tried adding bridge_early_media to the bridge - no effect. The problem is
> when the call  is picked during the the second bridge which is used to
> create a different ring on the extensions (there is a later bridge to
> answer) the other phones in the group continue to ring. I am trying to
> solve this which for some reason occurs more after I updated to the recent
> git. I am hoping if I can combine into a single bridge command the problem
> will be solved. Any suggestion greatly appreciated.
>
> BTW, the command below does nothing, nothing in the debug, it shows up but
> nothing happens. I tried using ! , | but no luck.
>
>    <action application="bridge" data="[originate_timeout=17]group/home@
> ${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>
>
>
> On Apr 4, 2012, at 10:51 AM, Mario G wrote:
>
> Testing the latest git has introduced an issue but I wonder if I can get
> around it by combining the snippet below into one bridge command. If I can
> set the ringback in the bridge it may fix it. I did not see this in the
> wiki. Can it be done?
> Mario G
>
>         <action application="bridge" data=
> "[originate_timeout=17]group/home@${domain_name}"/>
> <action application="set" data="ringback=${us-ring}"/>
> <action application="bridge" data="[originate_timeout=1]group/MGT@
> ${domain_name}"/>
>
> _________________________________________________________________________
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Thu Apr  5 01:39:28 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 14:39:28 -0700
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
Message-ID: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>

Is there a way to ring an internal user/group but not have it participate in the incoming call/bridge? I know sounds strange but I have a situation where I need to call a user group for 1 second (different ring on the phones) and proceeding in the extension dialplan. I tried the wiki on this one too. It has been dialed using bridge but that is causing problems if a call is picked up during that 1 ring. Thanks for any help.
Mario G

From gabe at gundy.org  Thu Apr  5 01:46:58 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 15:46:58 -0600
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
In-Reply-To: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
References: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
Message-ID: <CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>

On Wed, Apr 4, 2012 at 3:39 PM, Mario G <mario_fs at mgtech.com> wrote:
> I know sounds strange but I have a situation where I need to call a user group for 1 second (different ring on the phones) and proceeding in the extension dialplan.

I have to wonder, is there a better way to do what you're trying to
do? Of course we don't know the answer without knowing what it is you
hope to gain by ringing them once.

I can think of a number of ways you might do this, but they're all hacks.

What are you hoping to do?


Best,
Gabe


From mario_fs at mgtech.com  Thu Apr  5 01:54:52 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 14:54:52 -0700
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
In-Reply-To: <CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
Message-ID: <0EFEC6B3-4F9E-4E00-A6C5-7C86CF293BF9@mgtech.com>

It complicated but has been working fine since 2010 except for 1 little issue. I tried to keep the question simple but...  Here is what happens:

1. Incoming call, I set early media "Hi so and so..." followed by us-ring.
2. Bridge the call to internal group A rings for 17 seconds, if time out:
3. Internal group B rings for 1 second, same phones but different extension on them which causes a different ring that let's us know we are about to call cellphones.
4. Bridge call to internal phones again and 2 cell phones. If time out:
5. Play message and go to voicemail.

The problem is step 3. If the phone is picked up right in the 1 second the call is missed and the other phones keep ringing. I thought to try to combine step 2 and 3 into a single bridge command it may work. But... in another post I just made I realized it would be solved if I could just ring users/group without bridging the call that would solve the issues and be better. Have not found anything on the wiki yet.

BTW, the reason for the reset of the early media us-ring is if I don't do it the caller hears the first second of the Hi message. Since you can't put ringback into the bridge dialstring (I tried) I have to keep the 2 steps separate, which is why now I would just like to figure out how to dial without bridging.


On Apr 4, 2012, at 2:33 PM, Michael Collins wrote:

> What's the plain language description of what you're trying to do? I think we're all a bit confused about exactly which problem you're attempting to solve. A plain language description (without any telephonese) will definitely get us all on the same page. :)
> 
> Thanks,
> MC
> 
> On Wed, Apr 4, 2012 at 1:21 PM, Mario G <mario_fs at mgtech.com> wrote:
> I tried putting the ringback into a bridge command - no effect, I also tried adding bridge_early_media to the bridge - no effect. The problem is when the call  is picked during the the second bridge which is used to create a different ring on the extensions (there is a later bridge to answer) the other phones in the group continue to ring. I am trying to solve this which for some reason occurs more after I updated to the recent git. I am hoping if I can combine into a single bridge command the problem will be solved. Any suggestion greatly appreciated.
> 
> BTW, the command below does nothing, nothing in the debug, it shows up but nothing happens. I tried using ! , | but no luck.
> 
>    <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>
> 
> 
> On Apr 4, 2012, at 10:51 AM, Mario G wrote:
> 
>> Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done?
>> Mario G
>> 
>>         <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
>> 	<action application="set" data="ringback=${us-ring}"/>
>> 	<action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>
>> 
>> _________________________________________________________________________
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Thu Apr  5 02:22:54 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 15:22:54 -0700
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
In-Reply-To: <CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
References: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
	<CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
Message-ID: <E955D414-2F99-42C4-B23D-276E0C37EF68@mgtech.com>

I posted this then MC responded to Can ringback be variable in bridge command?. See that thread for full description but basically, if no one answers a call (bridge timeout) I bridge again to internal users and cell phones. It's before the second bridge that I ring another set of users for 1 second, they are the same phones but setup to ring differently so we know the call is about to bridge to the cell phones. It's very handy but if someone picks up during that 1 ring then there a problem. Thanks for any help, it's a real head scratcher for me. Works fine except for this inconvenience which for some reason "seems" happening more since I updated FS.
Mario G

On Apr 4, 2012, at 2:46 PM, Gabriel Gunderson wrote:

> On Wed, Apr 4, 2012 at 3:39 PM, Mario G <mario_fs at mgtech.com> wrote:
>> I know sounds strange but I have a situation where I need to call a user group for 1 second (different ring on the phones) and proceeding in the extension dialplan.
> 
> I have to wonder, is there a better way to do what you're trying to
> do? Of course we don't know the answer without knowing what it is you
> hope to gain by ringing them once.
> 
> I can think of a number of ways you might do this, but they're all hacks.
> 
> What are you hoping to do?
> 
> 
> Best,
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Thu Apr  5 02:41:56 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 4 Apr 2012 18:41:56 -0400
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
In-Reply-To: <0EFEC6B3-4F9E-4E00-A6C5-7C86CF293BF9@mgtech.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
	<0EFEC6B3-4F9E-4E00-A6C5-7C86CF293BF9@mgtech.com>
Message-ID: <CAKV-jFze4U6joBPUP=EAd5gv5dT5saYvcne1G3OrfvmpPfQdBQ@mail.gmail.com>

can you do paging on these phones? i could think of something where you set
the bridge to auto answer, play a wav file, kill the call, and continue on
with your dialplan.

-BDF

On Wed, Apr 4, 2012 at 5:54 PM, Mario G <mario_fs at mgtech.com> wrote:

> It complicated but has been working fine since 2010 except for 1 little
> issue. I tried to keep the question simple but...  Here is what happens:
>
> 1. Incoming call, I set early media "Hi so and so..." followed by us-ring.
> 2. Bridge the call to internal group A rings for 17 seconds, if time out:
> 3. Internal group B rings for 1 second, same phones but different
> extension on them which causes a different ring that let's us know we are
> about to call cellphones.
> 4. Bridge call to internal phones again and 2 cell phones. If time out:
> 5. Play message and go to voicemail.
>
> The problem is step 3. If the phone is picked up right in the 1 second the
> call is missed and the other phones keep ringing. I thought to try to
> combine step 2 and 3 into a single bridge command it may work. But... in
> another post I just made I realized it would be solved if I could just ring
> users/group without bridging the call that would solve the issues and be
> better. Have not found anything on the wiki yet.
>
> BTW, the reason for the reset of the early media us-ring is if I don't do
> it the caller hears the first second of the Hi message. Since you can't put
> ringback into the bridge dialstring (I tried) I have to keep the 2 steps
> separate, which is why now I would just like to figure out how to dial
> without bridging.
>
>
> On Apr 4, 2012, at 2:33 PM, Michael Collins wrote:
>
> What's the plain language description of what you're trying to do? I think
> we're all a bit confused about exactly which problem you're attempting to
> solve. A plain language description (without any telephonese) will
> definitely get us all on the same page. :)
>
> Thanks,
> MC
>
> On Wed, Apr 4, 2012 at 1:21 PM, Mario G <mario_fs at mgtech.com> wrote:
>
>> I tried putting the ringback into a bridge command - no effect, I also
>> tried adding bridge_early_media to the bridge - no effect. The problem is
>> when the call  is picked during the the second bridge which is used to
>> create a different ring on the extensions (there is a later bridge to
>> answer) the other phones in the group continue to ring. I am trying to
>> solve this which for some reason occurs more after I updated to the recent
>> git. I am hoping if I can combine into a single bridge command the problem
>> will be solved. Any suggestion greatly appreciated.
>>
>> BTW, the command below does nothing, nothing in the debug, it shows up
>> but nothing happens. I tried using ! , | but no luck.
>>
>>    <action application="bridge" data="[originate_timeout=17]group/home@
>> ${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>
>>
>>
>> On Apr 4, 2012, at 10:51 AM, Mario G wrote:
>>
>> Testing the latest git has introduced an issue but I wonder if I can get
>> around it by combining the snippet below into one bridge command. If I can
>> set the ringback in the bridge it may fix it. I did not see this in the
>> wiki. Can it be done?
>> Mario G
>>
>>         <action application="bridge" data=
>> "[originate_timeout=17]group/home@${domain_name}"/>
>>  <action application="set" data="ringback=${us-ring}"/>
>>  <action application="bridge" data="[originate_timeout=1]group/MGT@
>> ${domain_name}"/>
>>
>> _________________________________________________________________________
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
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From mario_fs at mgtech.com  Thu Apr  5 02:48:22 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 15:48:22 -0700
Subject: [Freeswitch-users] Can ringback be variable in bridge command?
In-Reply-To: <CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
Message-ID: <2CEC40DF-18EF-4B6C-BD56-F028D4ED9946@mgtech.com>

Added the majority of the dialplan... It complicated but has been working fine since 2010 except for 1 little issue. I tried to keep the question simple but...  Here is what happens:

1. Incoming call, I set early media "Hi so and so..." followed by us-ring.
2. Bridge the call to internal group A rings for 17 seconds, if time out:
3. Internal group B rings for 1 second, same phones but different extension on them which causes a different ring that let's us know we are about to call cellphones.
4. Bridge call to internal phones again and 2 cell phones. If time out:
5. Play message and go to voicemail.

The problem is step 3. If the phone is picked up right in the 1 second the call is missed and the other phones keep ringing. I thought to try to combine step 2 and 3 into a single bridge command it may work. But... in another post I just made I realized it would be solved if I could just ring users/group without bridging the call that would solve the issues and be better. Have not found anything on the wiki yet.

BTW, the reason for the reset of the early media us-ring is if I don't do it the caller hears the first second of the Hi message. You can't put ringback into the bridge dialstring (I tried) I have to keep the 2 steps separate, which is why now I would just like to figure out how to dial without bridging but wonder what would happen when call is picked up during then.

    <extension name="xxphone">
      <condition field="destination_number" expression="^2920$">
        <!-- Next lines allow control after bridge timeout and hangup after bridge -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>

        <!--  While ringing inside caller will hear "hi name" then standard ring -->  
	<action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>
        <!-- ring home extensions only for ## seconds -->
        <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
        <!-- ring different extension once so we know we're about to go to the second stage -->
        <!-- ringback must be reset otherwise caller will hear a first second of previous early media for this 1 ring -->
	<action application="set" data="ringback=${us-ring}"/>
        <action application="bridge" data="[originate_timeout=1,ringback=${us-ring}]group/MGT@${domain_name}"/>

        <!-- Say we are keep trying.... and setup so they hear music while ringing a second time-->
	<!-- Using mod_file_string we can keep the phones ringing while playing sounds! -->
	<action application="set" data="ringback=file_string://${xxsounds}hold_on.wav!${sayname}!${xxsounds}we_trying_please_hold.wav!local_stream://moh"/>
        <!-- this is needed to allow call_timeout to work after bridging to a gateway -->
        <action application="set" data="ignore_early_media=true"/>
        <!-- ring home extensions +cell phone for ## seconds  NOTE THE NEXT 2 LINES DID NOT WORK TO SET TIMEOUT -->
        <!-- NOTE: The use of :_: invokes enterprise syntax to ring simoultaneously!   ATT takes 8-10 rings before cell phone rings! -->
        <!-- DIAL_GATEWAY is set in the public folder dialplans  -->
        <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444:_:sofia/gateway/${dial_gateway}/19161112222"/>
        <!-- Timeout/No answer go to voicemail -->
        <!--action application="playback" data="${xxsounds}leave_a_message.wav"/-->
        <action application="playback" data="file_string://${xxsounds}please_leave_a_message.wav!//${sayname}"/>
        <action application="answer"/>
        <!-- Next line removes default greeting, we already played one above-->
        <action application="set" data="skip_greeting=true"/>
        <action application="voicemail" data="default ${domain_name} 100"/>
      </condition>
    </extension>

On Apr 4, 2012, at 2:33 PM, Michael Collins wrote:

> What's the plain language description of what you're trying to do? I think we're all a bit confused about exactly which problem you're attempting to solve. A plain language description (without any telephonese) will definitely get us all on the same page. :)
> 
> Thanks,
> MC
> 
> On Wed, Apr 4, 2012 at 1:21 PM, Mario G <mario_fs at mgtech.com> wrote:
> I tried putting the ringback into a bridge command - no effect, I also tried adding bridge_early_media to the bridge - no effect. The problem is when the call  is picked during the the second bridge which is used to create a different ring on the extensions (there is a later bridge to answer) the other phones in the group continue to ring. I am trying to solve this which for some reason occurs more after I updated to the recent git. I am hoping if I can combine into a single bridge command the problem will be solved. Any suggestion greatly appreciated.
> 
> BTW, the command below does nothing, nothing in the debug, it shows up but nothing happens. I tried using ! , | but no luck.
> 
>    <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>
> 
> 
> On Apr 4, 2012, at 10:51 AM, Mario G wrote:
> 
>> Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done?
>> Mario G
>> 
>>         <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
>> 	<action application="set" data="ringback=${us-ring}"/>
>> 	<action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>
>> 
>> _________________________________________________________________________
> 
> 
> _________________________________________________________________________
> Pr
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From msc at freeswitch.org  Thu Apr  5 02:57:41 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 Apr 2012 15:57:41 -0700
Subject: [Freeswitch-users] Terminal Emulators
Message-ID: <CAKzWOxWdZxpnJ0-CtQtSZZpa=gYTCnYRq1U-n6pq-rCr08UYKw@mail.gmail.com>

Hey folks,

Today on the conference call we had a brief but passionate discussion about
terminal emulators. I'm finding that more and more I'm using all of the
"big 3" operating systems and that they each have their own quirks and
peculiarities. I decided to start a wiki page on the topic:

http://wiki.freeswitch.org/wiki/Terminal_emulators

I have been spending more time than I really want to on a Windows laptop so
I have been getting familiar with putty. I added some putty info to the
wiki plus a few other tidbits. I would like to invite everyone to hop on
and add whatever knowledge or tips & tricks that they've learned over the
years. If you know of a good terminal program that you use then feel free
to add it to the list.

Thanks!

-MC
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From msc at freeswitch.org  Thu Apr  5 03:03:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 Apr 2012 16:03:55 -0700
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
In-Reply-To: <CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
References: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
	<CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
Message-ID: <CAKzWOxWYU-treqfkwkrkPB19XVDaqFBxhsWWbOG7K5a6j0dMBw@mail.gmail.com>

Well, some would argue that what he's doing now is a hack, so maybe a
better hack is in order. :)

Perhaps the 1-second bridge could be accomplished by executing an extension
that runs an api in a new thread that does a 1-second ring. That way, if
someone answers the 1 second call it isn't THE call, if you get my meaning.

Other ideas?

-MC

On Wed, Apr 4, 2012 at 2:46 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Wed, Apr 4, 2012 at 3:39 PM, Mario G <mario_fs at mgtech.com> wrote:
> > I know sounds strange but I have a situation where I need to call a user
> group for 1 second (different ring on the phones) and proceeding in the
> extension dialplan.
>
> I have to wonder, is there a better way to do what you're trying to
> do? Of course we don't know the answer without knowing what it is you
> hope to gain by ringing them once.
>
> I can think of a number of ways you might do this, but they're all hacks.
>
> What are you hoping to do?
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Thu Apr  5 03:12:54 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 4 Apr 2012 16:12:54 -0700
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
In-Reply-To: <CAKzWOxWYU-treqfkwkrkPB19XVDaqFBxhsWWbOG7K5a6j0dMBw@mail.gmail.com>
References: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
	<CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
	<CAKzWOxWYU-treqfkwkrkPB19XVDaqFBxhsWWbOG7K5a6j0dMBw@mail.gmail.com>
Message-ID: <DAF28D60-EB35-4266-AA9B-230B0E261B73@mgtech.com>

I added the dialplan ext to the other post. Too bad I can't do a originate_timeout=0 that would cause the user to ring once but nothing else. I will look into the api call but was hoping for something simpler.
Mario G

On Apr 4, 2012, at 4:03 PM, Michael Collins wrote:

> Well, some would argue that what he's doing now is a hack, so maybe a better hack is in order. :)
> 
> Perhaps the 1-second bridge could be accomplished by executing an extension that runs an api in a new thread that does a 1-second ring. That way, if someone answers the 1 second call it isn't THE call, if you get my meaning. 
> 
> Other ideas?
> 
> -MC
> 
> On Wed, Apr 4, 2012 at 2:46 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Wed, Apr 4, 2012 at 3:39 PM, Mario G <mario_fs at mgtech.com> wrote:
> > I know sounds strange but I have a situation where I need to call a user group for 1 second (different ring on the phones) and proceeding in the extension dialplan.
> 
> I have to wonder, is there a better way to do what you're trying to
> do? Of course we don't know the answer without knowing what it is you
> hope to gain by ringing them once.
> 
> I can think of a number of ways you might do this, but they're all hacks.
> 
> What are you hoping to do?
> 
> 
> Best,
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr  5 03:19:43 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 5 Apr 2012 02:19:43 +0300
Subject: [Freeswitch-users] Is it possible to ring user without bridging?
In-Reply-To: <DAF28D60-EB35-4266-AA9B-230B0E261B73@mgtech.com>
References: <E27499FC-0602-4F4F-B394-D1B6F549BA87@mgtech.com>
	<CAMwZ-t+jG64Aw6J27_R_O3QPSqaLTBs44Y=cigLj=S4-ZcP+ag@mail.gmail.com>
	<CAKzWOxWYU-treqfkwkrkPB19XVDaqFBxhsWWbOG7K5a6j0dMBw@mail.gmail.com>
	<DAF28D60-EB35-4266-AA9B-230B0E261B73@mgtech.com>
Message-ID: <CALNZuwmzfcxwwN0Do_0z1F=RJ9151rDENXi2LqWkDLcnB=hNKA@mail.gmail.com>

Maybe... you could set
group_confirm_key<http://wiki.freeswitch.org/wiki/Variable_group_confirm_key>
so
even if they pick up accidentally they still need to press 1.
-Avi


On Thu, Apr 5, 2012 at 2:12 AM, Mario G <mario_fs at mgtech.com> wrote:

> I added the dialplan ext to the other post. Too bad I can't do a
> originate_timeout=0 that would cause the user to ring once but nothing
> else. I will look into the api call but was hoping for something simpler.
> Mario G
>
>
> On Apr 4, 2012, at 4:03 PM, Michael Collins wrote:
>
> Well, some would argue that what he's doing now is a hack, so maybe a
> better hack is in order. :)
>
> Perhaps the 1-second bridge could be accomplished by executing an
> extension that runs an api in a new thread that does a 1-second ring. That
> way, if someone answers the 1 second call it isn't THE call, if you get my
> meaning.
>
> Other ideas?
>
> -MC
>
> On Wed, Apr 4, 2012 at 2:46 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Wed, Apr 4, 2012 at 3:39 PM, Mario G <mario_fs at mgtech.com> wrote:
>> > I know sounds strange but I have a situation where I need to call a
>> user group for 1 second (different ring on the phones) and proceeding in
>> the extension dialplan.
>>
>> I have to wonder, is there a better way to do what you're trying to
>> do? Of course we don't know the answer without knowing what it is you
>> hope to gain by ringing them once.
>>
>> I can think of a number of ways you might do this, but they're all hacks.
>>
>> What are you hoping to do?
>>
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gabe at gundy.org  Thu Apr  5 05:21:18 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 19:21:18 -0600
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
Message-ID: <CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>

On Wed, Apr 4, 2012 at 9:29 AM, babak yakhchali
<babak.freeswitch at gmail.com> wrote:
> I'm trying to setup a centralized dialplan for ?multiple instances of
> freeswitch. I tried
> <X-PRE-PROCESS cmd="exec" data="wget -qO -
> http://www.example.com/dialplan.xml" />
> but it is not implemented on windows yet. is there any similar alternative?
> thanx

Isn't this what you're looking for?

http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22dialplan.22


Best,
Gabe


From gabe at gundy.org  Thu Apr  5 05:26:30 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 4 Apr 2012 19:26:30 -0600
Subject: [Freeswitch-users] help with using mod_xml_curl
In-Reply-To: <4f7c1e85.0517440a.6c9c.210e@mx.google.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8C2@RSV16.realitysolutions.local>
	<4f7c1e85.0517440a.6c9c.210e@mx.google.com>
Message-ID: <CAMwZ-tJASB6S7QFTd3mrOt9UdTyuU2FsyyGwKf+RkRzNH2-9nA@mail.gmail.com>

On Wed, Apr 4, 2012 at 4:17 AM, Ashish Lal <ashishlal.work at gmail.com> wrote:
> Freeswitch then queries a webserver using xml_curl to authenticate the
> username and session id. Is this possible? Can we use mod_xml_curl to send a
> session id? If so how?

This should get you started:
http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22


> Also, I am using a simple TCP server for a dummy webserver. Do XML responses
> from the TCP server have to be wrapped in HTTP POST?

It's FreeSWITCH doing an HTTP POST to a web server and then the web
server responding with XML over HTTP. You're going to love it for its
power and flexibility.

Good luck, happy hacking and welcome to FreeSWITCH!



Best,
Gabe


From mitch.capper at gmail.com  Thu Apr  5 05:50:24 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 4 Apr 2012 18:50:24 -0700
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
Message-ID: <CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>

While cygwin definitely works its overkill, there are fantastic
windows builds of wget and curl with full SSL support, if you have an
issue finding them let us know but I use them on a daily basis.

~Mitch


From babak.freeswitch at gmail.com  Thu Apr  5 11:24:15 2012
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 5 Apr 2012 11:54:15 +0430
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
	<CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>
Message-ID: <CAEVfcaY0UzYsN4QoF-D2W1okg18kfq5zHcXxJixB+j6FymXpvw@mail.gmail.com>

thanks for answering.
But my real problem is not wget itself. I know that there is a version of
wget for windows. My real problem is the freeswitch <X-PRE-PROCESS
cmd="exec"... . If you use it on windows build of freeswitch, you see that
it is not implemented on windows yet. I wanted to know if there is any
workaround to implement something like the solution suggested in
http://wiki.freeswitch.org/wiki/Mod_xml_curl#Storing_Static_Dialplans
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From lorentz.nails at gmail.com  Thu Apr  5 13:40:52 2012
From: lorentz.nails at gmail.com (lorentz.nails)
Date: Thu, 5 Apr 2012 11:40:52 +0200
Subject: [Freeswitch-users] 3-user conference from 2-user call
Message-ID: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>

I would like to create a 3-user conference from a 2-user call (bridge)
without calling the third user twice.

The scenario is the following:
1. user 1000 calls user 1001 (that responds);
2. user 1001 may decide to create a 3-user conference by transferring the
execution to an extension that use conference_set_auto_outcall to invite
user 1002 and create the conference.

<action application="bind_meta_app" data="3 b s execute_extension::cf XML
features"/>

<extension name="cf">
    <condition field="destination_number" expression="^cf$">
        <action application="answer"/>
        <action application="transfer" data="-both 3000 XML default"/>
    </condition>
</extension>

<extension name="nb_conferences">
    <condition field="destination_number" expression="^(3000)$">
        <action application="answer"/>
        <action application="set"
data="conference_auto_outcall_timeout=20"/>
        <action application="set"
data="conference_auto_outcall_flags=none"/>
        <action application="set"
data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/>
        <action application="set"
data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>
        <action application="set"
data="conference_auto_outcall_profile=default"/>
        <action application="conference_set_auto_outcall" data="user/1002@
${domain}"/>
        <action application="conference" data="$1 at default"/>
    </condition>
</extension>

At this point user 1002 receive two simultaneous calls: one from user 1000
and one from user 1001 (I'm using softphones).

The behavior I would like to achieve is that user 1002 only gets one call
from user 1001 or from the conference.


Lorentz
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From curriegrad2004 at gmail.com  Thu Apr  5 18:14:48 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 5 Apr 2012 07:14:48 -0700
Subject: [Freeswitch-users] 3-user conference from 2-user call
In-Reply-To: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>
References: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>
Message-ID: <CAMWiMy9YBMXg=iDJg-s6Piu=XCsRkfPdg34tZzMnRcLDfdR5Mw@mail.gmail.com>

My approach for what you're trying to do would be:
1. Transfer user 1000 and 1001 to a conference bridge
2. Use the originate command to call 1002 and once he answers, have
him to be placed in the conference

On Thu, Apr 5, 2012 at 2:40 AM, lorentz.nails <lorentz.nails at gmail.com> wrote:
> I would like to create a 3-user conference from a 2-user call (bridge)
> without calling the third user twice.
>
> The scenario is the following:
> 1. user 1000 calls user 1001 (that responds);
> 2. user 1001 may decide to create a 3-user conference by transferring the
> execution to an extension that use conference_set_auto_outcall to invite
> user 1002 and create the conference.
>
> <action application="bind_meta_app" data="3 b s execute_extension::cf XML
> features"/>
>
> <extension name="cf">
> ??? <condition field="destination_number" expression="^cf$">
> ??? ??? <action application="answer"/>
> ??? ??? <action application="transfer" data="-both 3000 XML default"/>
> ??? </condition>
> </extension>
>
> <extension name="nb_conferences">
> ??? <condition field="destination_number" expression="^(3000)$">
> ??? ??? <action application="answer"/>
> ??? ??? <action application="set"
> data="conference_auto_outcall_timeout=20"/>
> ??? ??? <action application="set"
> data="conference_auto_outcall_flags=none"/>
> ??? ??? <action application="set"
> data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/>
> ??? ??? <action application="set"
> data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>
> ??? ??? <action application="set"
> data="conference_auto_outcall_profile=default"/>
> ??? ??? <action application="conference_set_auto_outcall"
> data="user/1002@${domain}"/>
> ??? ??? <action application="conference" data="$1 at default"/>
> ??? </condition>
> </extension>
>
> At this point user 1002 receive two simultaneous calls: one from user 1000
> and one from user 1001 (I'm using softphones).
>
> The behavior I would like to achieve is that user 1002 only gets one call
> from user 1001 or from the conference.
>
>
> Lorentz
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From avi at avimarcus.net  Thu Apr  5 18:18:28 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 5 Apr 2012 17:18:28 +0300
Subject: [Freeswitch-users] 3-user conference from 2-user call
In-Reply-To: <CAMWiMy9YBMXg=iDJg-s6Piu=XCsRkfPdg34tZzMnRcLDfdR5Mw@mail.gmail.com>
References: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>
	<CAMWiMy9YBMXg=iDJg-s6Piu=XCsRkfPdg34tZzMnRcLDfdR5Mw@mail.gmail.com>
Message-ID: <CALNZuwmx=KNELgKf1pHBtSc2NRVTXSZTAVhcseBx3rYZ-DC2pw@mail.gmail.com>

Maybe you could use an attended transfer and then press 0 to turn it into a
three-way conference:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer

-Avi



On Thu, Apr 5, 2012 at 5:14 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> My approach for what you're trying to do would be:
> 1. Transfer user 1000 and 1001 to a conference bridge
> 2. Use the originate command to call 1002 and once he answers, have
> him to be placed in the conference
>
> On Thu, Apr 5, 2012 at 2:40 AM, lorentz.nails <lorentz.nails at gmail.com>
> wrote:
> > I would like to create a 3-user conference from a 2-user call (bridge)
> > without calling the third user twice.
> >
> > The scenario is the following:
> > 1. user 1000 calls user 1001 (that responds);
> > 2. user 1001 may decide to create a 3-user conference by transferring the
> > execution to an extension that use conference_set_auto_outcall to invite
> > user 1002 and create the conference.
> >
> > <action application="bind_meta_app" data="3 b s execute_extension::cf XML
> > features"/>
> >
> > <extension name="cf">
> >     <condition field="destination_number" expression="^cf$">
> >         <action application="answer"/>
> >         <action application="transfer" data="-both 3000 XML default"/>
> >     </condition>
> > </extension>
> >
> > <extension name="nb_conferences">
> >     <condition field="destination_number" expression="^(3000)$">
> >         <action application="answer"/>
> >         <action application="set"
> > data="conference_auto_outcall_timeout=20"/>
> >         <action application="set"
> > data="conference_auto_outcall_flags=none"/>
> >         <action application="set"
> >
> data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/>
> >         <action application="set"
> >
> data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>
> >         <action application="set"
> > data="conference_auto_outcall_profile=default"/>
> >         <action application="conference_set_auto_outcall"
> > data="user/1002@${domain}"/>
> >         <action application="conference" data="$1 at default"/>
> >     </condition>
> > </extension>
> >
> > At this point user 1002 receive two simultaneous calls: one from user
> 1000
> > and one from user 1001 (I'm using softphones).
> >
> > The behavior I would like to achieve is that user 1002 only gets one call
> > from user 1001 or from the conference.
> >
> >
> > Lorentz
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From yehavi.bourvine at gmail.com  Thu Apr  5 18:23:49 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 5 Apr 2012 17:23:49 +0300
Subject: [Freeswitch-users] 3-user conference from 2-user call
In-Reply-To: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>
References: <CAOU8kFfr5z5tZGp4r18=8V9V3ZnK3RzOcsw0m5xn-w040spcMw@mail.gmail.com>
Message-ID: <CAAtWKeE+d8Qwkde5emzObBBg0pvEa7j8dDxRLfKc3pPBh2p4Dg@mail.gmail.com>

If all you need is a 3-party call then why won't you do it via the phone's
confference option? As far as I know all IP phones support 3 way call, some
of them even support 4 and 5 way call.

                         __Yehavi:


2012/4/5 lorentz.nails <lorentz.nails at gmail.com>

> I would like to create a 3-user conference from a 2-user call (bridge)
> without calling the third user twice.
>
> The scenario is the following:
> 1. user 1000 calls user 1001 (that responds);
> 2. user 1001 may decide to create a 3-user conference by transferring the
> execution to an extension that use conference_set_auto_outcall to invite
> user 1002 and create the conference.
>
> <action application="bind_meta_app" data="3 b s execute_extension::cf XML
> features"/>
>
> <extension name="cf">
>     <condition field="destination_number" expression="^cf$">
>         <action application="answer"/>
>         <action application="transfer" data="-both 3000 XML default"/>
>     </condition>
> </extension>
>
> <extension name="nb_conferences">
>     <condition field="destination_number" expression="^(3000)$">
>         <action application="answer"/>
>         <action application="set"
> data="conference_auto_outcall_timeout=20"/>
>         <action application="set"
> data="conference_auto_outcall_flags=none"/>
>         <action application="set"
> data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/>
>         <action application="set"
> data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>
>         <action application="set"
> data="conference_auto_outcall_profile=default"/>
>         <action application="conference_set_auto_outcall" data="user/1002@
> ${domain}"/>
>         <action application="conference" data="$1 at default"/>
>     </condition>
> </extension>
>
> At this point user 1002 receive two simultaneous calls: one from user 1000
> and one from user 1001 (I'm using softphones).
>
> The behavior I would like to achieve is that user 1002 only gets one call
> from user 1001 or from the conference.
>
>
> Lorentz
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Adam.Lappe at qsc.de  Thu Apr  5 17:55:08 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Thu, 5 Apr 2012 15:55:08 +0200
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>

Hi all,

i am trying to dynamically set the mod-sound variable in mod_conference, so I can play different music on hold
to my conference rooms without having to create multiple profiles.

Is there a way to set my moh-sound path while I create the conference?

Or can I change the moh-sound file via the event socket in a existing conference?
I think this would also lead to success.

Thanks in advance,
Adam

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From azza.miled at gmail.com  Thu Apr  5 18:26:35 2012
From: azza.miled at gmail.com (azza miled)
Date: Thu, 5 Apr 2012 15:26:35 +0100
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAGQdkzQXGm8J3VsR3q8WZBeMRgYsa4aJgG9RDEbOLi4DwqC1Mw@mail.gmail.com>

I am trying to install FS in a debian machin. I usd the tutoriel on
wiki.When I type make, I obtain this error msg
cc1: all warnings being treated as errors
make[9]: *** [nua_subnotref.lo] Erreur 1
make[8]: *** [all] Erreur 2
make[8]: *** Pas de r?gle pour fabriquer la cible ? nua/libnua.la ?,
n?cessaire pour ? libsofia-sip-ua.la ?. Arr?t.
make[7]: *** [all-recursive] Erreur 1
Making all in packages
make[6]: *** [all-recursive] Erreur 1
make[5]: *** [all] Erreur 2
make[4]: *** [/usr/local/src/FreeSWITCH/libs/sofia-sip/libsofia-sip-ua/
libsofia-sip-ua.la]
Erreur 2
make[3]: *** [mod_sofia-all] Erreur 1
make[2]: *** [all-recursive] Erreur 1
make[1]: *** [all-recursive] Erreur 1
make: *** [all] Erreur 2
Can any one help, please
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From msc at freeswitch.org  Thu Apr  5 20:36:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Apr 2012 09:36:20 -0700
Subject: [Freeswitch-users] collect dtmf events for a timed period
In-Reply-To: <20120403184336.27640@gmx.com>
References: <20120403184336.27640@gmx.com>
Message-ID: <CAKzWOxUutb_epwZ9NTkwuTdUY_BTGLThH9wPqxzp2AN89mifYQ@mail.gmail.com>

You can use setInputCallback. I covered this pretty well in the FreeSWITCH
"bridge" book in chapter 7. There's also a bit of documentation on the wiki
here: http://wiki.freeswitch.org/wiki/Lua#session:setInputCallback

-MC

On Tue, Apr 3, 2012 at 11:43 AM, adam harris <admharris at gmx.com> wrote:

> How can I set to recevice dtmf events for a timed period in lua. I want to
> evaluate each key press so this requires consuming events or callback I
> would guess. Any sugesstions or examples
>
>
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From krice at freeswitch.org  Thu Apr  5 20:37:39 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 05 Apr 2012 11:37:39 -0500
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAGQdkzQXGm8J3VsR3q8WZBeMRgYsa4aJgG9RDEbOLi4DwqC1Mw@mail.gmail.com>
Message-ID: <CBA33484.30DAD%krice@freeswitch.org>

That?s only part of the error message.... Git pull, boostrap, configure, and
make again... 

If you checked out in a window 14 to 8 hours before the timestamp on this
email its possible you got part of some changes that were going in for zrtp

K



On 4/5/12 9:26 AM, "azza miled" <azza.miled at gmail.com> wrote:

> I am trying to install FS in a debian machin. I usd the tutoriel on wiki.When
> I type make, I obtain this error msg
> cc1: all warnings being treated as errors
> make[9]: *** [nua_subnotref.lo] Erreur 1
> make[8]: *** [all] Erreur 2
> make[8]: *** Pas de r?gle pour fabriquer la cible ? nua/libnua.la
> <http://libnua.la/> ??,
> n?cessaire pour ??libsofia-sip-ua.la <http://libsofia-sip-ua.la/> ??. Arr?t.
> make[7]: *** [all-recursive] Erreur 1
> Making all in packages
> make[6]: *** [all-recursive] Erreur 1
> make[5]: *** [all] Erreur 2
> make[4]: *** 
> [/usr/local/src/FreeSWITCH/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la
> <http://libsofia-sip-ua.la/> ]
> Erreur 2
> make[3]: *** [mod_sofia-all] Erreur 1
> make[2]: *** [all-recursive] Erreur 1
> make[1]: *** [all-recursive] Erreur 1
> make: *** [all] Erreur 2
> Can any one help, please
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From david.villasmil.work at gmail.com  Thu Apr  5 20:42:48 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 5 Apr 2012 18:42:48 +0200
Subject: [Freeswitch-users] FreeSWITCH-Billing
In-Reply-To: <CAKMOEZ=HSXrU5fS9vvc0qYz+z9Qj+M-j53+OhUdWXtMQ2i-c7g@mail.gmail.com>
References: <CAKMOEZkEinT7FBLQQuMS2x9tuGs8E5NmgCDKQGnvs-JvQO+cLw@mail.gmail.com>
	<CAFGRPVof4tLBSakAv_VBFY8ptufL5OSF8Nrh0QfexJ19aV2KaQ@mail.gmail.com>
	<CAKMOEZk3OqJ1Om1J_E7wr8OiQYzqPf22axf++hO0xHuewH3bFw@mail.gmail.com>
	<CAFGRPVorc822vyJW-k-8DY6JsRjAEkV49YxTEg0_E_KQk8mbaw@mail.gmail.com>
	<CAKMOEZ=HSXrU5fS9vvc0qYz+z9Qj+M-j53+OhUdWXtMQ2i-c7g@mail.gmail.com>
Message-ID: <CAFGRPVrJJOSDwijaU2YSd_zszf+1WFs8Nh+X8OL-P7KHJY9iGA@mail.gmail.com>

Hello Dihn,

Again, read the readme. You need to set the variables accordingly. You need
to have apache2 running and set the www directory. you also need to run
that as root.

David

On Wed, Apr 4, 2012 at 6:43 PM, dinh thuc <dinhthuc1303 at gmail.com> wrote:

> Hi David !!
>
> Thank for quick reply ! I have install mysql success. But I have other
> error when I copy file
>
>
> ERROR 1227 (42000) at line 553: Access denied; you need the SUPER
> privilege for this operation
>
> copying scripts: cp -r /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway/*
> /usr/local/freeswitch/
> copying scripts: cp -r
> /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway_scripts/*
> /home/freeswitch/
> Creating default Sofia configuration
>
> Couldn't get sofia.conf.xml! at ./install.pl line 127.
>
> I can copy manual file ?
>
> B&R
>
> Dinh Thuc
>
>
> V?o 21:34 Ng?y 04 th?ng 4 n?m 2012, David Villasmil <
> david.villasmil.work at gmail.com> ?? vi?t:
>
> Dihn,
>>
>> You need to read the readme. You must have mysql installed.
>>
>> David
>>
>>
>> On Wed, Apr 4, 2012 at 12:39 PM, dinh thuc <dinhthuc1303 at gmail.com>wrote:
>>
>>> I have run "install.pl" but I have some error !!
>>>
>>> sh: mysql: command not found
>>> sh: mysql: command not found
>>> sh: mysql: command not found
>>> copying scripts: cp -r
>>> /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway/* /usr/local/freeswitch/
>>> cp: target `/usr/local/freeswitch/' is not a directory
>>> copying scripts: cp -r
>>> /root/davidcsi-FreeSWITCH-Billing-f079e0f/gateway_scripts/* /home/david/
>>> cp: target `/home/david/' is not a directory
>>> Creating default Sofia configuration
>>> /usr/local/freeswitch//conf/autoload_configs/sofia.conf.xml: No such
>>> file or directory
>>> Couldn't get sofia.conf.xml! at ./install.pl line 127.
>>>
>>>
>>> B&R
>>>
>>> Dinh Thuc
>>>
>>> V?o 17:26 Ng?y 04 th?ng 4 n?m 2012, David Villasmil <
>>> david.villasmil.work at gmail.com> ?? vi?t:
>>>
>>> Hello Dihn,
>>>>
>>>> There is a script called "install.pl", you need to execute that.
>>>>
>>>> David
>>>>
>>>>
>>>> On Wed, Apr 4, 2012 at 7:02 AM, dinh thuc <dinhthuc1303 at gmail.com>wrote:
>>>>
>>>>> Hi David !!
>>>>>
>>>>> We need to solution about billing in my system. I have read link
>>>>> https://github.com/davidcsi/FreeSWITCH-Billing
>>>>>
>>>>> You have script for install all ?
>>>>>
>>>>> I want to install and test
>>>>>
>>>>> B&R
>>>>>
>>>>> Dinh Thuc
>>>>>
>>>>
>>>>
>>>
>>
>
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From msc at freeswitch.org  Thu Apr  5 20:48:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Apr 2012 09:48:58 -0700
Subject: [Freeswitch-users] Playing additional file when transferring
	call
In-Reply-To: <8220BA48EF3C9A438124F20C9C76571520EE6782@srv01.khdev.corp>
References: <8220BA48EF3C9A438124F20C9C76571520EE6782@srv01.khdev.corp>
Message-ID: <CAKzWOxW3tvKLXtvbp=_fS4C5wRAzpKmGuefvEsX-8TTOjoXNmA@mail.gmail.com>

How is the transfer executed? Transfer button on the phone or with the DTMF
controls in the Local_Extension?
-MC

On Mon, Apr 2, 2012 at 11:54 AM, Klaus Hochlehnert <Mailings at kh-dev.de>wrote:

>  Hi,****
>
> ** **
>
> I have a question if this is possible and how:****
>
> When transferring a call (but only when it was picked up and from an
> external source) I?d like to play a complete sound file till the end before
> going to ?transfer_ringback?.****
>
> ** **
>
> Does anyone have an idea how to do that?****
>
> ** **
>
> Thanks****
>
> Klaus****
>
>
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From mario_fs at mgtech.com  Thu Apr  5 21:06:40 2012
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 5 Apr 2012 10:06:40 -0700
Subject: [Freeswitch-users] [Solved] Can ringback be variable in bridge
	command?
In-Reply-To: <2CEC40DF-18EF-4B6C-BD56-F028D4ED9946@mgtech.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
	<2CEC40DF-18EF-4B6C-BD56-F028D4ED9946@mgtech.com>
Message-ID: <F06AE99E-B32D-4E34-9501-0CA2598BA9D3@mgtech.com>

For those of you that tried to help here is my solution. I tried various things but found I could send an alert-info to the SPA962s, once that worked I removed the second bridge to ring a different tone via different group. Now, I just export an alert-info prior the second bridge that causes a different internal ring to let us know cell phones are also being rung. Neato!

On Apr 4, 2012, at 3:48 PM, Mario G wrote:

> Added the majority of the dialplan... It complicated but has been working fine since 2010 except for 1 little issue. I tried to keep the question simple but...  Here is what happens:
> 
> 1. Incoming call, I set early media "Hi so and so..." followed by us-ring.
> 2. Bridge the call to internal group A rings for 17 seconds, if time out:
> 3. Internal group B rings for 1 second, same phones but different extension on them which causes a different ring that let's us know we are about to call cellphones.
> 4. Bridge call to internal phones again and 2 cell phones. If time out:
> 5. Play message and go to voicemail.
> 
> The problem is step 3. If the phone is picked up right in the 1 second the call is missed and the other phones keep ringing. I thought to try to combine step 2 and 3 into a single bridge command it may work. But... in another post I just made I realized it would be solved if I could just ring users/group without bridging the call that would solve the issues and be better. Have not found anything on the wiki yet.
> 
> BTW, the reason for the reset of the early media us-ring is if I don't do it the caller hears the first second of the Hi message. You can't put ringback into the bridge dialstring (I tried) I have to keep the 2 steps separate, which is why now I would just like to figure out how to dial without bridging but wonder what would happen when call is picked up during then.
> 
>     <extension name="xxphone">
>       <condition field="destination_number" expression="^2920$">
>         <!-- Next lines allow control after bridge timeout and hangup after bridge -->
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="continue_on_fail=true"/>
> 
>         <!--  While ringing inside caller will hear "hi name" then standard ring -->  
> 	<action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>
>         <!-- ring home extensions only for ## seconds -->
>         <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
>         <!-- ring different extension once so we know we're about to go to the second stage -->
>         <!-- ringback must be reset otherwise caller will hear a first second of previous early media for this 1 ring -->
> 	<action application="set" data="ringback=${us-ring}"/>
>         <action application="bridge" data="[originate_timeout=1,ringback=${us-ring}]group/MGT@${domain_name}"/>
> 
>         <!-- Say we are keep trying.... and setup so they hear music while ringing a second time-->
> 	<!-- Using mod_file_string we can keep the phones ringing while playing sounds! -->
> 	<action application="set" data="ringback=file_string://${xxsounds}hold_on.wav!${sayname}!${xxsounds}we_trying_please_hold.wav!local_stream://moh"/>
>         <!-- this is needed to allow call_timeout to work after bridging to a gateway -->
>         <action application="set" data="ignore_early_media=true"/>
>         <!-- ring home extensions +cell phone for ## seconds  NOTE THE NEXT 2 LINES DID NOT WORK TO SET TIMEOUT -->
>         <!-- NOTE: The use of :_: invokes enterprise syntax to ring simoultaneously!   ATT takes 8-10 rings before cell phone rings! -->
>         <!-- DIAL_GATEWAY is set in the public folder dialplans  -->
>         <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444:_:sofia/gateway/${dial_gateway}/19161112222"/>
>         <!-- Timeout/No answer go to voicemail -->
>         <!--action application="playback" data="${xxsounds}leave_a_message.wav"/-->
>         <action application="playback" data="file_string://${xxsounds}please_leave_a_message.wav!//${sayname}"/>
>         <action application="answer"/>
>         <!-- Next line removes default greeting, we already played one above-->
>         <action application="set" data="skip_greeting=true"/>
>         <action application="voicemail" data="default ${domain_name} 100"/>
>       </condition>
>     </extension>
> 
> On Apr 4, 2012, at 2:33 PM, Michael Collins wrote:
> 
>> What's the plain language description of what you're trying to do? I think we're all a bit confused about exactly which problem you're attempting to solve. A plain language description (without any telephonese) will definitely get us all on the same page. :)
>> 
>> Thanks,
>> MC
>> 
>> On Wed, Apr 4, 2012 at 1:21 PM, Mario G <mario_fs at mgtech.com> wrote:
>> I tried putting the ringback into a bridge command - no effect, I also tried adding bridge_early_media to the bridge - no effect. The problem is when the call  is picked during the the second bridge which is used to create a different ring on the extensions (there is a later bridge to answer) the other phones in the group continue to ring. I am trying to solve this which for some reason occurs more after I updated to the recent git. I am hoping if I can combine into a single bridge command the problem will be solved. Any suggestion greatly appreciated.
>> 
>> BTW, the command below does nothing, nothing in the debug, it shows up but nothing happens. I tried using ! , | but no luck.
>> 
>>    <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}![originate_timeout=1]group/MGT@${domain_name}"/>
>> 
>> 
>> On Apr 4, 2012, at 10:51 AM, Mario G wrote:
>> 
>>> Testing the latest git has introduced an issue but I wonder if I can get around it by combining the snippet below into one bridge command. If I can set the ringback in the bridge it may fix it. I did not see this in the wiki. Can it be done?
>>> Mario G
>>> 
>>>         <action application="bridge" data="[originate_timeout=17]group/home@${domain_name}"/>
>>> 	<action application="set" data="ringback=${us-ring}"/>
>>> 	<action application="bridge" data="[originate_timeout=1]group/MGT@${domain_name}"/>
>>> 
>>> _________________________________________________________________________
>> 
>> 
>> _________________________________________________________________________
>> Pr
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Thu Apr  5 21:10:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Apr 2012 10:10:55 -0700
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
Message-ID: <CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>

I have not seen a way to do this dynamically other than to use xml_curl and
serve up a conf profile with a custom moh-sound param.
-MC

On Thu, Apr 5, 2012 at 6:55 AM, Lappe, Adam <Adam.Lappe at qsc.de> wrote:

>  Hi all,****
>
> ** **
>
> i am trying to dynamically set the mod-sound variable in mod_conference,
> so I can play different music on hold****
>
> to my conference rooms without having to create multiple profiles.****
>
> ** **
>
> Is there a way to set my moh-sound path while I create the conference?****
>
> ** **
>
> Or can I change the moh-sound file via the event socket in a existing
> conference?****
>
> I think this would also lead to success.****
>
> ** **
>
> Thanks in advance,****
>
> Adam****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Thu Apr  5 21:11:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Apr 2012 10:11:58 -0700
Subject: [Freeswitch-users] [Solved] Can ringback be variable in bridge
	command?
In-Reply-To: <F06AE99E-B32D-4E34-9501-0CA2598BA9D3@mgtech.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
	<2CEC40DF-18EF-4B6C-BD56-F028D4ED9946@mgtech.com>
	<F06AE99E-B32D-4E34-9501-0CA2598BA9D3@mgtech.com>
Message-ID: <CAKzWOxWamakxEhp_SK9Bh5WOu4z7orsM89ZBx3NDB1fKwYq8Kw@mail.gmail.com>

So what you're saying is that the SPA962 can do something other than sit
around and take up space? :)
-MC

On Thu, Apr 5, 2012 at 10:06 AM, Mario G <mario_fs at mgtech.com> wrote:

> For those of you that tried to help here is my solution. I tried various
> things but found I could send an alert-info to the SPA962s, once that
> worked I removed the second bridge to ring a different tone via different
> group. Now, I just export an alert-info prior the second bridge that causes
> a different internal ring to let us know cell phones are also being rung.
> Neato!
>
>
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From mthakershi at gmail.com  Thu Apr  5 21:14:18 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 5 Apr 2012 12:14:18 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
Message-ID: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>

Hello,

Is there a way to disable core codecs offered by CORE_PCM_MODULE?

I want to use G729 as I need to make more concurrent calls with limited
bandwidth. But no matter what I do, system keeps using PCMU codec.

I did following modifications based on the wiki but nothing seems to work.

In modules.conf.xml,
<load module="mod_g729"/>

Under SIP profile file,
<param name="inbound-codec-negotiation" value="greedy"/>

Under vars.xml file,
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>

I also contacted my SIP provider (Vitelity) but they say if I disable other
codecs completely, I may be able to use G729.

Thanks for help.

Malay Thakershi
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From mario_fs at mgtech.com  Thu Apr  5 21:29:40 2012
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 5 Apr 2012 10:29:40 -0700
Subject: [Freeswitch-users] [Solved] Can ringback be variable in bridge
	command?
In-Reply-To: <CAKzWOxWamakxEhp_SK9Bh5WOu4z7orsM89ZBx3NDB1fKwYq8Kw@mail.gmail.com>
References: <C9F2DE62-B47D-4FD9-904E-690AD36E89B2@mgtech.com>
	<ECA1D731-6D8B-4520-84EB-5CD0152070BF@mgtech.com>
	<CAKzWOxXzoRxw-d8fa9ZdtT6sg4cy-0YVW4-NeUygM3S3d5AfrQ@mail.gmail.com>
	<2CEC40DF-18EF-4B6C-BD56-F028D4ED9946@mgtech.com>
	<F06AE99E-B32D-4E34-9501-0CA2598BA9D3@mgtech.com>
	<CAKzWOxWamakxEhp_SK9Bh5WOu4z7orsM89ZBx3NDB1fKwYq8Kw@mail.gmail.com>
Message-ID: <0346815B-3E9C-45B3-A7D2-D2F50439EA87@mgtech.com>

I gather you don't like the SPAs, they were my first SIP phones, would never buy Cisco/Linksys again. Been looking for something to replace them for a while now. Thinking Snom...

On Apr 5, 2012, at 10:11 AM, Michael Collins wrote:

> So what you're saying is that the SPA962 can do something other than sit around and take up space? :)
> -MC
> 
> On Thu, Apr 5, 2012 at 10:06 AM, Mario G <mario_fs at mgtech.com> wrote:
> For those of you that tried to help here is my solution. I tried various things but found I could send an alert-info to the SPA962s, once that worked I removed the second bridge to ring a different tone via different group. Now, I just export an alert-info prior the second bridge that causes a different internal ring to let us know cell phones are also being rung. Neato!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Thu Apr  5 21:43:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Apr 2012 10:43:11 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
Message-ID: <CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>

can you give us a debug log of a call doing this?
-MC

On Thu, Apr 5, 2012 at 10:14 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Hello,
>
> Is there a way to disable core codecs offered by CORE_PCM_MODULE?
>
> I want to use G729 as I need to make more concurrent calls with limited
> bandwidth. But no matter what I do, system keeps using PCMU codec.
>
> I did following modifications based on the wiki but nothing seems to work.
>
> In modules.conf.xml,
> <load module="mod_g729"/>
>
> Under SIP profile file,
> <param name="inbound-codec-negotiation" value="greedy"/>
>
> Under vars.xml file,
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729"/>
>
> I also contacted my SIP provider (Vitelity) but they say if I disable
> other codecs completely, I may be able to use G729.
>
> Thanks for help.
>
> Malay Thakershi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From david.villasmil.work at gmail.com  Thu Apr  5 22:26:40 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 5 Apr 2012 20:26:40 +0200
Subject: [Freeswitch-users] originate
Message-ID: <CAFGRPVpmJB69KHs5JJFf+6zydAmm=50X2Z+S5v5gXFmAJDUViQ@mail.gmail.com>

Hello guys,

i'm trying to execute an originate

originate sofia/external/669448337 at 1.2.3.4 0013058883456
+OK a27c762c-7f4a-11e1-9269-f52ca8b46747

2012-04-05 20:10:32.893284 [ERR] switch_core_io.c:929 Codec PROXY
PASS-THROUGH encoder error!

WHY do i see that error? codecs are fine both on FS and and the endpoint...

I originally want to originate and then generate dtmf to the leg... if
anyone can give me a help, i would appreciate it

thanks!

David
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From gcd at i.ph  Thu Apr  5 22:55:49 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 6 Apr 2012 02:55:49 +0800
Subject: [Freeswitch-users] originate
In-Reply-To: <CAFGRPVpmJB69KHs5JJFf+6zydAmm=50X2Z+S5v5gXFmAJDUViQ@mail.gmail.com>
References: <CAFGRPVpmJB69KHs5JJFf+6zydAmm=50X2Z+S5v5gXFmAJDUViQ@mail.gmail.com>
Message-ID: <CAHKqMb4sTV17+gqSsmmQA=He5SWc3LBx8C51W94a6pJDhVGHhQ@mail.gmail.com>

check if this is what you
need:http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf<%20http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf>

On Fri, Apr 6, 2012 at 2:26 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> i'm trying to execute an originate
>
> originate sofia/external/669448337 at 1.2.3.4 0013058883456
> +OK a27c762c-7f4a-11e1-9269-f52ca8b46747
>
> 2012-04-05 20:10:32.893284 [ERR] switch_core_io.c:929 Codec PROXY
> PASS-THROUGH encoder error!
>
> WHY do i see that error? codecs are fine both on FS and and the endpoint...
>
> I originally want to originate and then generate dtmf to the leg... if
> anyone can give me a help, i would appreciate it
>
> thanks!
>
> David
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr  5 22:58:27 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 5 Apr 2012 21:58:27 +0300
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
Message-ID: <CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>

Try absolute_codec_string<http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
-Avi
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From david.villasmil.work at gmail.com  Thu Apr  5 23:44:54 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 5 Apr 2012 21:44:54 +0200
Subject: [Freeswitch-users] originate
In-Reply-To: <CAHKqMb4sTV17+gqSsmmQA=He5SWc3LBx8C51W94a6pJDhVGHhQ@mail.gmail.com>
References: <CAFGRPVpmJB69KHs5JJFf+6zydAmm=50X2Z+S5v5gXFmAJDUViQ@mail.gmail.com>
	<CAHKqMb4sTV17+gqSsmmQA=He5SWc3LBx8C51W94a6pJDhVGHhQ@mail.gmail.com>
Message-ID: <CAFGRPVrUGk8_1bacd7ETxqJk6gZR+eVDTBx3aHT5j98Z6afHjw@mail.gmail.com>

Hello nandy,

Yep, I've seen that. I was wondering about the error i got when originating.

David

On Thu, Apr 5, 2012 at 8:55 PM, Nandy Dagondon <gcd at i.ph> wrote:

> check if this is what you need:http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_queue_dtmf
>
> On Fri, Apr 6, 2012 at 2:26 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> i'm trying to execute an originate
>>
>> originate sofia/external/669448337 at 1.2.3.4 0013058883456
>> +OK a27c762c-7f4a-11e1-9269-f52ca8b46747
>>
>> 2012-04-05 20:10:32.893284 [ERR] switch_core_io.c:929 Codec PROXY
>> PASS-THROUGH encoder error!
>>
>> WHY do i see that error? codecs are fine both on FS and and the
>> endpoint...
>>
>> I originally want to originate and then generate dtmf to the leg... if
>> anyone can give me a help, i would appreciate it
>>
>> thanks!
>>
>> David
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Thu Apr  5 23:48:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Apr 2012 14:48:15 -0500
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAEVfcaY0UzYsN4QoF-D2W1okg18kfq5zHcXxJixB+j6FymXpvw@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
	<CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>
	<CAEVfcaY0UzYsN4QoF-D2W1okg18kfq5zHcXxJixB+j6FymXpvw@mail.gmail.com>
Message-ID: <CAKbxfG_A7TyAMpDD+x=g4yFYWkD2PxHQ5p635Q3BPU-Z8S8JiA@mail.gmail.com>

We need someone who knows the windows way to do that to code in the support.
You can always use mod_xml_curl or mod_lua with the xml binding hooks



On Thu, Apr 5, 2012 at 2:24 AM, babak yakhchali
<babak.freeswitch at gmail.com> wrote:
> thanks for answering.
> But my real problem is not wget itself. I know that there is a version of
> wget for windows. My real problem is the freeswitch?<X-PRE-PROCESS
> cmd="exec"... . If you use it on windows build of freeswitch, you see that
> it is not implemented on windows yet. I wanted to know if there is any
> workaround to implement something like the solution suggested
> in?http://wiki.freeswitch.org/wiki/Mod_xml_curl#Storing_Static_Dialplans
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jaganthoutam at gmail.com  Fri Apr  6 05:08:00 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Fri, 6 Apr 2012 06:38:00 +0530
Subject: [Freeswitch-users] fs_cli error
Message-ID: <CAHqn9HEKU3XckUrbXvTXZS95ZX39wfHOcDt4FCELYFeUzuchBQ@mail.gmail.com>

I am getting below error when i try to fs_cli


freeswitch version :  FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05
14-28-01 -0500)



[root at localhost freeswitch]# ./fs_cli

[ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
Usage: ./fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
command] [-t <timeout_ms>] [profile]

  -?,-h --help                    Usage Information
  -H, --host=hostname             Host to connect
  -P, --port=port                 Port to connect (1 - 65535)
  -u, --user=user at domain          user at domain
  -p, --password=password         Password
  -i, --interrupt                 Allow Control-c to interrupt
  -x, --execute=command           Execute Command and Exit
  -l, --loglevel=command          Log Level
  -q, --quiet                     Disable logging
  -r, --retry                     Retry connection on failure
  -R, --reconnect                 Reconnect if disconnected
  -d, --debug=level               Debug Level (0 - 7)
  -t, --timeout                   Timeout for API commands (in miliseconds)
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From krice at freeswitch.org  Fri Apr  6 05:12:55 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 05 Apr 2012 20:12:55 -0500
Subject: [Freeswitch-users] fs_cli error
In-Reply-To: <CAHqn9HEKU3XckUrbXvTXZS95ZX39wfHOcDt4FCELYFeUzuchBQ@mail.gmail.com>
Message-ID: <CBA3AD47.30DD4%krice@freeswitch.org>

That means freeswitch isnt running or you changed how its listening for ESL
or you didn?t load mod_event_socket


On 4/5/12 8:08 PM, "Jagadish Thoutam" <jaganthoutam at gmail.com> wrote:

> I am getting below error when i try to fs_cli
> 
> 
> freeswitch version :? FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05
> 14-28-01 -0500)
> 
> 
> 
> [root at localhost freeswitch]# ./fs_cli
> 
> [ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
> Usage: ./fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [-t <timeout_ms>] [profile]
> 
> ? -?,-h --help??????????????????? Usage Information
> ? -H, --host=hostname???????????? Host to connect
> ? -P, --port=port???????????????? Port to connect (1 - 65535)
> ? -u, --user=user at domain????????? user at domain
> ? -p, --password=password???????? Password
> ? -i, --interrupt???????????????? Allow Control-c to interrupt
> ? -x, --execute=command?????????? Execute Command and Exit
> ? -l, --loglevel=command????????? Log Level
> ? -q, --quiet???????????????????? Disable logging
> ? -r, --retry???????????????????? Retry connection on failure
> ? -R, --reconnect???????????????? Reconnect if disconnected
> ? -d, --debug=level?????????????? Debug Level (0 - 7)
> ? -t, --timeout?????????????????? Timeout for API commands (in miliseconds)
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fs-list at communicatefreely.net  Fri Apr  6 06:56:51 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Thu, 05 Apr 2012 22:56:51 -0400
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
In-Reply-To: <CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
	<CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>
Message-ID: <4F7E5B73.5000004@communicatefreely.net>

xml_curl is worth the effort, as it lets you dynamically set anything in 
a conference profile - from sample rate, to default energy levels.  If 
you are using a database already, it's not a complicated script to 
write, and it lets you do all of that on the fly.

-Tim

Michael Collins wrote:
> I have not seen a way to do this dynamically other than to use 
> xml_curl and serve up a conf profile with a custom moh-sound param.
> -MC
>
> On Thu, Apr 5, 2012 at 6:55 AM, Lappe, Adam <Adam.Lappe at qsc.de 
> <mailto:Adam.Lappe at qsc.de>> wrote:
>
>     Hi all,
>
>      
>
>     i am trying to dynamically set the mod-sound variable in
>     mod_conference, so I can play different music on hold
>
>     to my conference rooms without having to create multiple profiles.
>
>      
>
>     Is there a way to set my moh-sound path while I create the conference?
>
>      
>
>     Or can I change the moh-sound file via the event socket in a
>     existing conference?
>
>     I think this would also lead to success.
>
>      
>
>     Thanks in advance,
>
>     Adam
>
>      
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anton.jugatsu at gmail.com  Fri Apr  6 08:47:14 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 6 Apr 2012 08:47:14 +0400
Subject: [Freeswitch-users] fs_cli error
In-Reply-To: <CBA3AD47.30DD4%krice@freeswitch.org>
References: <CAHqn9HEKU3XckUrbXvTXZS95ZX39wfHOcDt4FCELYFeUzuchBQ@mail.gmail.com>
	<CBA3AD47.30DD4%krice@freeswitch.org>
Message-ID: <CANBhyPCOoSgGU6vpuGJLeEJLzt+n7HMq-P6ZwUseZXH2WGDFrA@mail.gmail.com>

You can check running freeswtich by submitting this commonds:
pgrep -lf freeswitch
lsof -iudp -ac freeswitch

6 ?????? 2012 ?. 5:12 ???????????? Ken Rice <krice at freeswitch.org> ???????:

>  That means freeswitch isnt running or you changed how its listening for
> ESL or you didn't load mod_event_socket
>
>
>
> On 4/5/12 8:08 PM, "Jagadish Thoutam" <jaganthoutam at gmail.com> wrote:
>
> I am getting below error when i try to fs_cli
>
>
> freeswitch version :  FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05
> 14-28-01 -0500)
>
>
>
> [root at localhost freeswitch]# ./fs_cli
>
> [ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
> Usage: ./fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
> command] [-t <timeout_ms>] [profile]
>
>   -?,-h --help                    Usage Information
>   -H, --host=hostname             Host to connect
>   -P, --port=port                 Port to connect (1 - 65535)
>   -u, --user=user at domain          user at domain
>   -p, --password=password         Password
>   -i, --interrupt                 Allow Control-c to interrupt
>   -x, --execute=command           Execute Command and Exit
>   -l, --loglevel=command          Log Level
>   -q, --quiet                     Disable logging
>   -r, --retry                     Retry connection on failure
>   -R, --reconnect                 Reconnect if disconnected
>   -d, --debug=level               Debug Level (0 - 7)
>   -t, --timeout                   Timeout for API commands (in miliseconds)
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From babak.freeswitch at gmail.com  Fri Apr  6 11:20:15 2012
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Fri, 6 Apr 2012 11:50:15 +0430
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
 cmd="exec" ...
In-Reply-To: <CAKbxfG_A7TyAMpDD+x=g4yFYWkD2PxHQ5p635Q3BPU-Z8S8JiA@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
	<CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>
	<CAEVfcaY0UzYsN4QoF-D2W1okg18kfq5zHcXxJixB+j6FymXpvw@mail.gmail.com>
	<CAKbxfG_A7TyAMpDD+x=g4yFYWkD2PxHQ5p635Q3BPU-Z8S8JiA@mail.gmail.com>
Message-ID: <CAEVfcaYthVyFip_JUqTJtmg2noiHUUY3aiB0Rx3hmwy77mxNrg@mail.gmail.com>

Thanx Anthony
You are right and I'm trying xml_curl now. I hope someone implement "exec"
for windows
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From mytemike72 at gmail.com  Fri Apr  6 11:53:40 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Fri, 6 Apr 2012 09:53:40 +0200
Subject: [Freeswitch-users] FW: Inbound ESL .RecvEventTimed(0) Blocking?
Message-ID: <039801cd13ca$62fe58d0$28fb0a70$@com>

Hi,

 

Can someone please give me an answer, I am problaby not the first to notice.

 

I need to know why (and how to avoid)  ESLevent RecvEventTimed(0) being
blocking and keeps waiting for event to arrive,  as it should not.

It seems to ignore any form of time you pass as parameter and just always
wait for an event.

 

Btw, I am on Windows.

 

Thanks!

Mike.

 

From: Michael Lutz 
Sent: donderdag 5 april 2012 22:19
To: 'FreeSWITCH Users Help'
Subject: Inbound ESL .RecvEventTimed(0) Blocking?

 

Hi Guys,

 

Can anybody please help me, I am using .RecvEventTimed(0) on a inbound ESL
connection to check if events are there.

According to specs, this should be non-blocking and as the wiki says "used
for polling".

 

However, the code just seems to stop until it gets an event.

 

I am listening with a filter for a specific uuid like:

 

ESLevent eslEvent = fsConnection.SendRecv("event plain ALL");

eslEvent = fsConnection.SendRecv("filter Unique-ID " + thisCall.callId);

.

Some code.

.

eslEvent = fsConnection.RecvEventTimed(0);

 

But it just waits for an event on the RecvEventTimed(0);

When I generate an event (for example pressing a dtmf) is continues.

 

Is there some extra thing I need to do?

 

Thanks,

Mike.

 

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From jaganthoutam at gmail.com  Fri Apr  6 14:18:03 2012
From: jaganthoutam at gmail.com (Jagadish Thoutam)
Date: Fri, 6 Apr 2012 15:48:03 +0530
Subject: [Freeswitch-users] fs_cli error
In-Reply-To: <CANBhyPCOoSgGU6vpuGJLeEJLzt+n7HMq-P6ZwUseZXH2WGDFrA@mail.gmail.com>
References: <CAHqn9HEKU3XckUrbXvTXZS95ZX39wfHOcDt4FCELYFeUzuchBQ@mail.gmail.com>
	<CBA3AD47.30DD4%krice@freeswitch.org>
	<CANBhyPCOoSgGU6vpuGJLeEJLzt+n7HMq-P6ZwUseZXH2WGDFrA@mail.gmail.com>
Message-ID: <CAHqn9HELA8jUMZ3Rj_R9XWJzzrt5ozoofOu2C9Hwy=fymvGYzg@mail.gmail.com>

thanks got it working now.

On 6 April 2012 10:17, Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> You can check running freeswtich by submitting this commonds:
> pgrep -lf freeswitch
> lsof -iudp -ac freeswitch
>
> 6 ?????? 2012 ?. 5:12 ???????????? Ken Rice <krice at freeswitch.org>???????:
>
>  That means freeswitch isnt running or you changed how its listening for
>> ESL or you didn't load mod_event_socket
>>
>>
>>
>> On 4/5/12 8:08 PM, "Jagadish Thoutam" <jaganthoutam at gmail.com> wrote:
>>
>> I am getting below error when i try to fs_cli
>>
>>
>> freeswitch version :  FreeSWITCH Version 1.1.beta1 (git-c41a16d
>> 2012-04-05 14-28-01 -0500)
>>
>>
>>
>> [root at localhost freeswitch]# ./fs_cli
>>
>> [ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
>> Usage: ./fs_cli [-H <host>] [-P <port>] [-p <secret>] [-d <level>] [-x
>> command] [-t <timeout_ms>] [profile]
>>
>>   -?,-h --help                    Usage Information
>>   -H, --host=hostname             Host to connect
>>   -P, --port=port                 Port to connect (1 - 65535)
>>   -u, --user=user at domain          user at domain
>>   -p, --password=password         Password
>>   -i, --interrupt                 Allow Control-c to interrupt
>>   -x, --execute=command           Execute Command and Exit
>>   -l, --loglevel=command          Log Level
>>   -q, --quiet                     Disable logging
>>   -r, --retry                     Retry connection on failure
>>   -R, --reconnect                 Reconnect if disconnected
>>   -d, --debug=level               Debug Level (0 - 7)
>>   -t, --timeout                   Timeout for API commands (in
>> miliseconds)
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jeff at jefflenk.com  Fri Apr  6 19:04:18 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 6 Apr 2012 08:04:18 -0700 (PDT)
Subject: [Freeswitch-users] windows alternative for <X-PRE-PROCESS
	cmd="exec" ...
In-Reply-To: <CAEVfcaYthVyFip_JUqTJtmg2noiHUUY3aiB0Rx3hmwy77mxNrg@mail.gmail.com>
References: <CAEVfcabEhvZQEzLm=mwfePZ+7RFANabCMr3O2xK--VHps11hTA@mail.gmail.com>
	<CAMwZ-tLrisbKRTESNACQ-F_k0rpW0aqG-TYPt5VfcPgv78TDvg@mail.gmail.com>
	<CALn233pQzF0dkdg0XW8ot2oynBHuVT1TSgg0_SGgdahj3c17Bg@mail.gmail.com>
	<CAEVfcaY0UzYsN4QoF-D2W1okg18kfq5zHcXxJixB+j6FymXpvw@mail.gmail.com>
	<CAKbxfG_A7TyAMpDD+x=g4yFYWkD2PxHQ5p635Q3BPU-Z8S8JiA@mail.gmail.com>
	<CAEVfcaYthVyFip_JUqTJtmg2noiHUUY3aiB0Rx3hmwy77mxNrg@mail.gmail.com>
Message-ID: <1333724658326-7443244.post@n2.nabble.com>

This is a fairly large task and may not be 100%. There are some examples to
look at including tcsh for windows. If someone wants to take a stab at this
please do.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/windows-alternative-for-X-PRE-PROCESS-cmd-exec-tp7436900p7443244.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From acrow at integrafin.co.uk  Fri Apr  6 21:31:13 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 06 Apr 2012 18:31:13 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F78827F.7020401@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>,
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>,
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
Message-ID: <4F7F2861.1030007@integrafin.co.uk>

On 01/04/12 17:29, Alex Crow wrote:
> On 01/04/12 11:26, Peter Olsson wrote:
>> In your sample you're actually queuing "sleep(1000)" to the channel. 
>> So first it will queue play_and_get_digits, inside that loop it will 
>> find more commands in the queue, and exeute them, so in the end it 
>> will only execute a bunch of "sleep(1000)".
>>
>> You're better of using a sleep in your module directly (not tell FS 
>> to sleep). Also you will need to poll the events in the loop 
>> somehow., checkout the perl sample here: 
>> http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example
>
> Peter,
>
> I've changed to the below (adapted that Perl example) and I only hear 
> the prompt for digits when the webserver times out after 60 seconds. 
> Any time I make any call to ESL after the execute, I don't hear the 
> prompt in a timely fashion. This is so complicated as I'm not really 
> that good at software development, I'm more a systems/networks guy. I 
> feel like a total thickie here, but the docs don't seem to explain in 
> which order I should call things and expect to get replies, etc.
>
> $esl = new ESLconnection('localhost','8021','ClueCon');
>
> $event = $esl->events('plain', 'ALL');
> //$event = $esl->filter("Unique-ID",$uuid);
>
> sleep (1);
>
> $res = $esl->execute("play_and_get_digits","4 11 1 5000 # 
> ivr/ivr-please_enter_the_phone_number.wav 
> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");
>
> $job_uuid = $res->getHeader("Job-UUID");
> $status = $res->getHeader("Reply-Text");
>
>
>
> //$foo = $uuid . ' target_num';
>
> echo $status;
>
> $stay_connected = 1;
> while ( $stay_connected ) {
>     $e = $esl->recvEventTimed(0);
>     if ( $e ) {
>         $ev_name = $e->getHeader("Event-Name");
>         # Should we check for the $job_uuid to match the background job ?
>         //if ( $ev_name == 'BACKGROUND_JOB' ) {
>         //    if ( $e->getHeader("Job-UUID") == $uuid ) {
>         //        $call_result = $e->getBody();
>         //        echo "Result of call to $target was $call_result\n\n";
>         //    }
>         //} else
>         echo $ev_name;
>         if ( $ev_name == 'EXECUTE_COMPLETE' ) {
>             //my $digit = $e->getHeader("DTMF-Digit");
>            // print "Received DTMF digit: $digit\n";
>             //if ( $digit =~ m/\D/ ) {a
>              //   print "Exiting...\n";
>                 //if ( $e->getHeader("Job-UUID") == $job_uuid ) {
>                 $stay_connected = 0;
>                 //}
>             //}
>         } else {
>       # Some other event
>         }
>     } else {
>     # do other things while waiting for events
>         //sleep (1);
>     }
> }
>
> $event=$esl->sendRecv("api uuid_getvar $uuid target_num");
>
>
> $ret = $event->getBody();
>
> $esl->disconnect();
>
> Apologies for my newbishness and many thanks for your help so far.
>
> Hoping for an epiphany when I sudddenly understand what I am doing!
>
> Alex

Can anyone help with this? I updated to latest Git, thinking it may have 
been a bug, and still have the same problem, when I try to wait for 
events as in the above code, it stops the playback of the "enter the 
phone number" from happening until after the web server times out.

Logically I can't see a reason for this, is it because you can't use an 
inbound connection activated by calling from curl in the dialplan? If 
this is indeed the case I could try rewriting this for outbound but I 
feel that will be more complex for a relative beginner programmer.

If I have missed some vital or obvious concept feel free to chastise me :-).

Thanks

Alex



From anthony.minessale at gmail.com  Fri Apr  6 21:59:46 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Apr 2012 12:59:46 -0500
Subject: [Freeswitch-users] FW: Inbound ESL .RecvEventTimed(0) Blocking?
In-Reply-To: <039801cd13ca$62fe58d0$28fb0a70$@com>
References: <039801cd13ca$62fe58d0$28fb0a70$@com>
Message-ID: <CAKbxfG8XtpYpZ816XV84V2--=tm7++6Bv8VeUpOBCbNm5hxHEg@mail.gmail.com>

try 1 instead of 0 i believe 0 means forever

On Fri, Apr 6, 2012 at 2:53 AM, Michael Lutz <mytemike72 at gmail.com> wrote:
> Hi,
>
>
>
> Can someone please give me an answer, I am problaby not the first to notice?
>
>
>
> I need to know why (and how to avoid)? ESLevent RecvEventTimed(0) being
> blocking and keeps waiting for event to arrive,? as it should not.
>
> It seems to ignore any form of time you pass as parameter and just always
> wait for an event.
>
>
>
> Btw, I am on Windows?
>
>
>
> Thanks!
>
> Mike.
>
>
>
> From: Michael Lutz
> Sent: donderdag 5 april 2012 22:19
> To: 'FreeSWITCH Users Help'
> Subject: Inbound ESL .RecvEventTimed(0) Blocking?
>
>
>
> Hi Guys,
>
>
>
> Can anybody please help me, I am using .RecvEventTimed(0) on a inbound ESL
> connection to check if events are there.
>
> According to specs, this should be non-blocking and as the wiki says ?used
> for polling?.
>
>
>
> However, the code just seems to stop until it gets an event.
>
>
>
> I am listening with a filter for a specific uuid like:
>
>
>
> ESLevent eslEvent = fsConnection.SendRecv("event plain ALL");
>
> eslEvent = fsConnection.SendRecv("filter Unique-ID " + thisCall.callId);
>
> ?
>
> Some code?
>
> ?
>
> eslEvent = fsConnection.RecvEventTimed(0);
>
>
>
> But it just waits for an event on the RecvEventTimed(0);
>
> When I generate an event (for example pressing a dtmf) is continues?
>
>
>
> Is there some extra thing I need to do?
>
>
>
> Thanks,
>
> Mike.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Apr  6 22:19:02 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Apr 2012 13:19:02 -0500
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F7F2861.1030007@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
Message-ID: <CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>

I think you might be twisted around its possible that doing what you
are doing is much more complicated than just using the socket app.
If you are just trying to play a file and get a digits etc you could
do it all with existing dialplan apps.



On Fri, Apr 6, 2012 at 12:31 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
> On 01/04/12 17:29, Alex Crow wrote:
>> On 01/04/12 11:26, Peter Olsson wrote:
>>> In your sample you're actually queuing "sleep(1000)" to the channel.
>>> So first it will queue play_and_get_digits, inside that loop it will
>>> find more commands in the queue, and exeute them, so in the end it
>>> will only execute a bunch of "sleep(1000)".
>>>
>>> You're better of using a sleep in your module directly (not tell FS
>>> to sleep). Also you will need to poll the events in the loop
>>> somehow., checkout the perl sample here:
>>> http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example
>>
>> Peter,
>>
>> I've changed to the below (adapted that Perl example) and I only hear
>> the prompt for digits when the webserver times out after 60 seconds.
>> Any time I make any call to ESL after the execute, I don't hear the
>> prompt in a timely fashion. This is so complicated as I'm not really
>> that good at software development, I'm more a systems/networks guy. I
>> feel like a total thickie here, but the docs don't seem to explain in
>> which order I should call things and expect to get replies, etc.
>>
>> $esl = new ESLconnection('localhost','8021','ClueCon');
>>
>> $event = $esl->events('plain', 'ALL');
>> //$event = $esl->filter("Unique-ID",$uuid);
>>
>> sleep (1);
>>
>> $res = $esl->execute("play_and_get_digits","4 11 1 5000 #
>> ivr/ivr-please_enter_the_phone_number.wav
>> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");
>>
>> $job_uuid = $res->getHeader("Job-UUID");
>> $status = $res->getHeader("Reply-Text");
>>
>>
>>
>> //$foo = $uuid . ' target_num';
>>
>> echo $status;
>>
>> $stay_connected = 1;
>> while ( $stay_connected ) {
>> ? ? $e = $esl->recvEventTimed(0);
>> ? ? if ( $e ) {
>> ? ? ? ? $ev_name = $e->getHeader("Event-Name");
>> ? ? ? ? # Should we check for the $job_uuid to match the background job ?
>> ? ? ? ? //if ( $ev_name == 'BACKGROUND_JOB' ) {
>> ? ? ? ? // ? ?if ( $e->getHeader("Job-UUID") == $uuid ) {
>> ? ? ? ? // ? ? ? ?$call_result = $e->getBody();
>> ? ? ? ? // ? ? ? ?echo "Result of call to $target was $call_result\n\n";
>> ? ? ? ? // ? ?}
>> ? ? ? ? //} else
>> ? ? ? ? echo $ev_name;
>> ? ? ? ? if ( $ev_name == 'EXECUTE_COMPLETE' ) {
>> ? ? ? ? ? ? //my $digit = $e->getHeader("DTMF-Digit");
>> ? ? ? ? ? ?// print "Received DTMF digit: $digit\n";
>> ? ? ? ? ? ? //if ( $digit =~ m/\D/ ) {a
>> ? ? ? ? ? ? ?// ? print "Exiting...\n";
>> ? ? ? ? ? ? ? ? //if ( $e->getHeader("Job-UUID") == $job_uuid ) {
>> ? ? ? ? ? ? ? ? $stay_connected = 0;
>> ? ? ? ? ? ? ? ? //}
>> ? ? ? ? ? ? //}
>> ? ? ? ? } else {
>> ? ? ? # Some other event
>> ? ? ? ? }
>> ? ? } else {
>> ? ? # do other things while waiting for events
>> ? ? ? ? //sleep (1);
>> ? ? }
>> }
>>
>> $event=$esl->sendRecv("api uuid_getvar $uuid target_num");
>>
>>
>> $ret = $event->getBody();
>>
>> $esl->disconnect();
>>
>> Apologies for my newbishness and many thanks for your help so far.
>>
>> Hoping for an epiphany when I sudddenly understand what I am doing!
>>
>> Alex
>
> Can anyone help with this? I updated to latest Git, thinking it may have
> been a bug, and still have the same problem, when I try to wait for
> events as in the above code, it stops the playback of the "enter the
> phone number" from happening until after the web server times out.
>
> Logically I can't see a reason for this, is it because you can't use an
> inbound connection activated by calling from curl in the dialplan? If
> this is indeed the case I could try rewriting this for outbound but I
> feel that will be more complex for a relative beginner programmer.
>
> If I have missed some vital or obvious concept feel free to chastise me :-).
>
> Thanks
>
> Alex
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From nasida at live.ru  Fri Apr  6 22:47:09 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 6 Apr 2012 22:47:09 +0400
Subject: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
Message-ID: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>


Hello guys!
I have set odbc mysql for internal profile. Can I remove sofia_reg_internal.db? I try to delete it but have lost my reg. After restart, FS creates this file again. What is core odbc support? FS just copy the data from file  sofia_reg_internal.db  to mysql via odbc ?
Thanks. 		 	   		  
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From msc at freeswitch.org  Fri Apr  6 23:12:14 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 Apr 2012 12:12:14 -0700
Subject: [Freeswitch-users] FW: Inbound ESL .RecvEventTimed(0) Blocking?
In-Reply-To: <039801cd13ca$62fe58d0$28fb0a70$@com>
References: <039801cd13ca$62fe58d0$28fb0a70$@com>
Message-ID: <CAKzWOxW6jSTbjMw0up+JfTkbnKZQv_M4imZS3SZ=i-G9BrnoTQ@mail.gmail.com>

On Fri, Apr 6, 2012 at 12:53 AM, Michael Lutz <mytemike72 at gmail.com> wrote:

> Hi,****
>
> ** **
>
> Can someone please give me an answer, I am problaby not the first to
> notice?****
>
> ** **
>
> I need to know why (and how to avoid)  ESLevent RecvEventTimed(0) being
> blocking and keeps waiting for event to arrive,  as it should not.****
>
> It seems to ignore any form of time you pass as parameter and just always
> wait for an event.****
>
> ** **
>
> Btw, I am on Windows?
>

You say that no matter what value you put in for the argument that it
always waits for an event? On a Linux box I tested Tony's suggestion about
using 1 vs 0 and it worked. I just tested on latest git from 2 days ago and
I definitely have "polling" behavior when I use 1 but I have "wait forever"
behavior when I use 0. I suspect that the wiki is wrong. (Sadly, this means
that chapter 9, page 224 is also wrong.)

I'll try testing on Windows later today and see what I get. BTW, are you
building FS from scratch or are you using the weekly binaries?

-MC
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From nestor at tiendalinux.com  Fri Apr  6 23:13:43 2012
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Fri, 06 Apr 2012 14:13:43 -0500
Subject: [Freeswitch-users] core dump on sip profile, fs version 2012-04-05
Message-ID: <4F7F4067.7000805@tiendalinux.com>

Hi guys, i have a core dump generated on freeswitch everytime i restart 
a new profile i have created:

freeswitch at internal> version
FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05 14-28-01 -0500)

freeswitch at internal> sofia profile internal-voice flush_inbound_reg reboot
Socket interrupted, bye!
[ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
[INFO] fs_cli.c:1326 main() Retrying

The full gdb trace is here (thanks to moises silva for the tip):

http://pastebin.com/d0uRnDrs

I updated from FreeSWITCH Version 1.0.head (git-9be51d5 2012-01-21 
13-45-21 -0500) to this version thinking it will fix the problem, but 
the problem remains.

the profile internal-voice is just the same internal profile with the 
following things changed:

# diff internal.xml  internal-voice.xml
1c1
< <profile name="internal">
---
 > <profile name="internal-voice">
18c18
< <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
---
 > <X-PRE-PROCESS cmd="include" data="internal-voice/*.xml"/>
79c79
< <param name="context" value="public"/>
---
 > <param name="context" value="internal"/>
89c89
< <param name="rtp-ip" value="$${local_ip_v4}"/>
---
 > <param name="rtp-ip" value="$${voice_ip_v4}"/>
91c91
< <param name="sip-ip" value="$${local_ip_v4}"/>
---
 > <param name="sip-ip" value="$${voice_ip_v4}"/>

where voice_ip_v4 is defined in vars.xml as follows:

<X-PRE-PROCESS cmd="set" data="voice_ip_v4=172.17.0.20"/>
<X-PRE-PROCESS cmd="set" data="voice_net_v4=172.17.0.0/24"/>

voice_ip_v4 is no other thing than a vlan interface for voip phones.

By proof and error i found that if i rename from internal-voice to 
another name, i.e.voice.xml then the sip profile doesn't core dump agin, 
a name problem ? i know that is difficult to believe but that's what i 
am experiencing, before upgrading gdb show me a problem on sofia_glue.c

On another freeswitch server i still have the problem of the event trap 
it change without having changed the ip address, like the following:

2012-03-22 20:34:22.353275 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change 
detected
2012-03-22 20:34:22.353275 [INFO] mod_sofia.c:5156 IP change detected 
[]->[] [fe80::225:90ff:fe3a:2e1a]->[::1]

I don't use the internal-ipv6 profile at all, so maybe the best thing i 
can do is to disable it.

I am not sure what triggers the above behavior but when it has been 
triggered then the vlan profile i use for the phones can't be started 
again, if i do a : `netstat -unlp` the related port is in use by the 
freeswitch process but the profile is down.

it writes a lock on the profile:

2012-03-22 20:34:24.253327 [DEBUG] sofia.c:2121 Write lock internal

but no 'unlock' like it does with the other profiles.

So i finally moved all the phones to the data vlan and no problem.

Another problem i have found is when i do a `show calls` and there is no 
calls the system show some calls, i found this only one time.  No space 
problem, Any explanation ?

Overall i am happy with my first freeswitch experience, it keeps on 
processing  from 30 to 40 calls concurrently on business hours from sip 
devices and a sangoma pri card with echo cancelation in conjunction with 
a sangoma transcoder card and a supermicro server.

Having working with asterisk before and sick with locks on the queue 
function i think freeswitch matters the transition from Asterisk, i have 
some things to master but i am on the way ....

Slds.

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia



From msc at freeswitch.org  Fri Apr  6 23:21:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 Apr 2012 12:21:00 -0700
Subject: [Freeswitch-users] core dump on sip profile,
	fs version 2012-04-05
In-Reply-To: <4F7F4067.7000805@tiendalinux.com>
References: <4F7F4067.7000805@tiendalinux.com>
Message-ID: <CAKzWOxVe63=h4s6v-j_e_njnz_MFO7rfSb+54W_ddZyEHrYN5A@mail.gmail.com>

This is great information! Of course, it's in the wrong place. :) Please
open a ticket at jira.freeswitch.org.
-MC

On Fri, Apr 6, 2012 at 12:13 PM, Nestor A Diaz <nestor at tiendalinux.com>wrote:

> Hi guys, i have a core dump generated on freeswitch everytime i restart
> a new profile i have created:
>
> freeswitch at internal> version
> FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05 14-28-01 -0500)
>
> freeswitch at internal> sofia profile internal-voice flush_inbound_reg reboot
> Socket interrupted, bye!
> [ERROR] fs_cli.c:1320 main() Error Connecting [Socket Connection Error]
> [INFO] fs_cli.c:1326 main() Retrying
>
> The full gdb trace is here (thanks to moises silva for the tip):
>
> http://pastebin.com/d0uRnDrs
>
> I updated from FreeSWITCH Version 1.0.head (git-9be51d5 2012-01-21
> 13-45-21 -0500) to this version thinking it will fix the problem, but
> the problem remains.
>
> the profile internal-voice is just the same internal profile with the
> following things changed:
>
> # diff internal.xml  internal-voice.xml
> 1c1
> < <profile name="internal">
> ---
>  > <profile name="internal-voice">
> 18c18
> < <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
> ---
>  > <X-PRE-PROCESS cmd="include" data="internal-voice/*.xml"/>
> 79c79
> < <param name="context" value="public"/>
> ---
>  > <param name="context" value="internal"/>
> 89c89
> < <param name="rtp-ip" value="$${local_ip_v4}"/>
> ---
>  > <param name="rtp-ip" value="$${voice_ip_v4}"/>
> 91c91
> < <param name="sip-ip" value="$${local_ip_v4}"/>
> ---
>  > <param name="sip-ip" value="$${voice_ip_v4}"/>
>
> where voice_ip_v4 is defined in vars.xml as follows:
>
> <X-PRE-PROCESS cmd="set" data="voice_ip_v4=172.17.0.20"/>
> <X-PRE-PROCESS cmd="set" data="voice_net_v4=172.17.0.0/24"/>
>
> voice_ip_v4 is no other thing than a vlan interface for voip phones.
>
> By proof and error i found that if i rename from internal-voice to
> another name, i.e.voice.xml then the sip profile doesn't core dump agin,
> a name problem ? i know that is difficult to believe but that's what i
> am experiencing, before upgrading gdb show me a problem on sofia_glue.c
>
> On another freeswitch server i still have the problem of the event trap
> it change without having changed the ip address, like the following:
>
> 2012-03-22 20:34:22.353275 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change
> detected
> 2012-03-22 20:34:22.353275 [INFO] mod_sofia.c:5156 IP change detected
> []->[] [fe80::225:90ff:fe3a:2e1a]->[::1]
>
> I don't use the internal-ipv6 profile at all, so maybe the best thing i
> can do is to disable it.
>
> I am not sure what triggers the above behavior but when it has been
> triggered then the vlan profile i use for the phones can't be started
> again, if i do a : `netstat -unlp` the related port is in use by the
> freeswitch process but the profile is down.
>
> it writes a lock on the profile:
>
> 2012-03-22 20:34:24.253327 [DEBUG] sofia.c:2121 Write lock internal
>
> but no 'unlock' like it does with the other profiles.
>
> So i finally moved all the phones to the data vlan and no problem.
>
> Another problem i have found is when i do a `show calls` and there is no
> calls the system show some calls, i found this only one time.  No space
> problem, Any explanation ?
>
> Overall i am happy with my first freeswitch experience, it keeps on
> processing  from 30 to 40 calls concurrently on business hours from sip
> devices and a sangoma pri card with echo cancelation in conjunction with
> a sangoma transcoder card and a supermicro server.
>
> Having working with asterisk before and sick with locks on the queue
> function i think freeswitch matters the transition from Asterisk, i have
> some things to master but i am on the way ....
>
> Slds.
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From acrow at integrafin.co.uk  Fri Apr  6 23:21:44 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 06 Apr 2012 20:21:44 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
Message-ID: <4F7F4248.3040404@integrafin.co.uk>

On 06/04/12 19:19, Anthony Minessale wrote:
> I think you might be twisted around its possible that doing what you
> are doing is much more complicated than just using the socket app.
> If you are just trying to play a file and get a digits etc you could
> do it all with existing dialplan apps.

Anthony,

Thanks very much for replying: what I wanted it it to do is things like 
ask a user if he'd like to set a pin for hotdesking (stored in a 
database, initially sqlite and later mysql or postgres), to check that 
pin when doing a login that logs out his existing extensions (if they 
exist), and various other features later  such as creating their own 
conferences with a pin of their choosing.

I thought the easiest way for the first task (and possibly others) would 
just be to set an extension in the dialplan to call a PHP page on a 
webserver via a curl action, passing some session variables, which would 
then take over the control of the call. I've already have written all 
the stuff for logging in/out and automatically rebooting/provisioning 
both Polycom and Snom phones but have been asked to add some security so 
someone can't just move someone else's registration to another device.

If you think that approach is wrong/doomed to failure and explain what 
the socket app is (is that the thing where you attach a call from the 
dialplan to a TCP socket? Then I've have to learn about setting up a 
listening app in PHP (or any other language), rather than via the 
webserver, wouldn't I, and then try and figure our how to filter our all 
the events from all the other calls, know which event I want, act on it, 
etc)

I just think that what I'm doing should be theoretically possible (if 
not the most elegant way) and just can't figure out why if I leave out 
the whole polling for events bit then the message is played, but as soon 
as I try to capture events *after* I've sent the execute call I never 
hear the message.

Just completely confused and am loathe to have to rewrite my hotdesking 
stuff which so far is really simple, I've attached the whole PHP app 
code here so you can see what I've done (it's only the "action=test" 
calling "saySomething" that I have been asking about.

Ideally I'd like to become at least adept if not expert in FS within a 
year as that would be the time we would think about replacing our 
existing phone system.

Cheers

Alex

>
>
>
> On Fri, Apr 6, 2012 at 12:31 PM, Alex Crow<acrow at integrafin.co.uk>  wrote:
>> On 01/04/12 17:29, Alex Crow wrote:
>>> On 01/04/12 11:26, Peter Olsson wrote:
>>>> In your sample you're actually queuing "sleep(1000)" to the channel.
>>>> So first it will queue play_and_get_digits, inside that loop it will
>>>> find more commands in the queue, and exeute them, so in the end it
>>>> will only execute a bunch of "sleep(1000)".
>>>>
>>>> You're better of using a sleep in your module directly (not tell FS
>>>> to sleep). Also you will need to poll the events in the loop
>>>> somehow., checkout the perl sample here:
>>>> http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example
>>> Peter,
>>>
>>> I've changed to the below (adapted that Perl example) and I only hear
>>> the prompt for digits when the webserver times out after 60 seconds.
>>> Any time I make any call to ESL after the execute, I don't hear the
>>> prompt in a timely fashion. This is so complicated as I'm not really
>>> that good at software development, I'm more a systems/networks guy. I
>>> feel like a total thickie here, but the docs don't seem to explain in
>>> which order I should call things and expect to get replies, etc.
>>>
>>> $esl = new ESLconnection('localhost','8021','ClueCon');
>>>
>>> $event = $esl->events('plain', 'ALL');
>>> //$event = $esl->filter("Unique-ID",$uuid);
>>>
>>> sleep (1);
>>>
>>> $res = $esl->execute("play_and_get_digits","4 11 1 5000 #
>>> ivr/ivr-please_enter_the_phone_number.wav
>>> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+", "$uuid");
>>>
>>> $job_uuid = $res->getHeader("Job-UUID");
>>> $status = $res->getHeader("Reply-Text");
>>>
>>>
>>>
>>> //$foo = $uuid . ' target_num';
>>>
>>> echo $status;
>>>
>>> $stay_connected = 1;
>>> while ( $stay_connected ) {
>>>      $e = $esl->recvEventTimed(0);
>>>      if ( $e ) {
>>>          $ev_name = $e->getHeader("Event-Name");
>>>          # Should we check for the $job_uuid to match the background job ?
>>>          //if ( $ev_name == 'BACKGROUND_JOB' ) {
>>>          //    if ( $e->getHeader("Job-UUID") == $uuid ) {
>>>          //        $call_result = $e->getBody();
>>>          //        echo "Result of call to $target was $call_result\n\n";
>>>          //    }
>>>          //} else
>>>          echo $ev_name;
>>>          if ( $ev_name == 'EXECUTE_COMPLETE' ) {
>>>              //my $digit = $e->getHeader("DTMF-Digit");
>>>             // print "Received DTMF digit: $digit\n";
>>>              //if ( $digit =~ m/\D/ ) {a
>>>               //   print "Exiting...\n";
>>>                  //if ( $e->getHeader("Job-UUID") == $job_uuid ) {
>>>                  $stay_connected = 0;
>>>                  //}
>>>              //}
>>>          } else {
>>>        # Some other event
>>>          }
>>>      } else {
>>>      # do other things while waiting for events
>>>          //sleep (1);
>>>      }
>>> }
>>>
>>> $event=$esl->sendRecv("api uuid_getvar $uuid target_num");
>>>
>>>
>>> $ret = $event->getBody();
>>>
>>> $esl->disconnect();
>>>
>>> Apologies for my newbishness and many thanks for your help so far.
>>>
>>> Hoping for an epiphany when I sudddenly understand what I am doing!
>>>
>>> Alex
>> Can anyone help with this? I updated to latest Git, thinking it may have
>> been a bug, and still have the same problem, when I try to wait for
>> events as in the above code, it stops the playback of the "enter the
>> phone number" from happening until after the web server times out.
>>
>> Logically I can't see a reason for this, is it because you can't use an
>> inbound connection activated by calling from curl in the dialplan? If
>> this is indeed the case I could try rewriting this for outbound but I
>> feel that will be more complex for a relative beginner programmer.
>>
>> If I have missed some vital or obvious concept feel free to chastise me :-).
>>
>> Thanks
>>
>> Alex
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From msc at freeswitch.org  Fri Apr  6 23:34:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 Apr 2012 12:34:22 -0700
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F7F4248.3040404@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
Message-ID: <CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>

Alex,

I noticed that you are using the same thing as Michael Lutz from the other
thread we just discussed. Could you try recvEventTimed(1) instead of
recvEventTimed(0)? I'm curious to see if it changes the behavior of your
application at all.

-MC

On Fri, Apr 6, 2012 at 12:21 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> On 06/04/12 19:19, Anthony Minessale wrote:
>
>> I think you might be twisted around its possible that doing what you
>> are doing is much more complicated than just using the socket app.
>> If you are just trying to play a file and get a digits etc you could
>> do it all with existing dialplan apps.
>>
>
> Anthony,
>
> Thanks very much for replying: what I wanted it it to do is things like
> ask a user if he'd like to set a pin for hotdesking (stored in a database,
> initially sqlite and later mysql or postgres), to check that pin when doing
> a login that logs out his existing extensions (if they exist), and various
> other features later  such as creating their own conferences with a pin of
> their choosing.
>
> I thought the easiest way for the first task (and possibly others) would
> just be to set an extension in the dialplan to call a PHP page on a
> webserver via a curl action, passing some session variables, which would
> then take over the control of the call. I've already have written all the
> stuff for logging in/out and automatically rebooting/provisioning both
> Polycom and Snom phones but have been asked to add some security so someone
> can't just move someone else's registration to another device.
>
> If you think that approach is wrong/doomed to failure and explain what the
> socket app is (is that the thing where you attach a call from the dialplan
> to a TCP socket? Then I've have to learn about setting up a listening app
> in PHP (or any other language), rather than via the webserver, wouldn't I,
> and then try and figure our how to filter our all the events from all the
> other calls, know which event I want, act on it, etc)
>
> I just think that what I'm doing should be theoretically possible (if not
> the most elegant way) and just can't figure out why if I leave out the
> whole polling for events bit then the message is played, but as soon as I
> try to capture events *after* I've sent the execute call I never hear the
> message.
>
> Just completely confused and am loathe to have to rewrite my hotdesking
> stuff which so far is really simple, I've attached the whole PHP app code
> here so you can see what I've done (it's only the "action=test" calling
> "saySomething" that I have been asking about.
>
> Ideally I'd like to become at least adept if not expert in FS within a
> year as that would be the time we would think about replacing our existing
> phone system.
>
> Cheers
>
> Alex
>
>
>>
>>
>> On Fri, Apr 6, 2012 at 12:31 PM, Alex Crow<acrow at integrafin.co.uk>
>>  wrote:
>>
>>> On 01/04/12 17:29, Alex Crow wrote:
>>>
>>>> On 01/04/12 11:26, Peter Olsson wrote:
>>>>
>>>>> In your sample you're actually queuing "sleep(1000)" to the channel.
>>>>> So first it will queue play_and_get_digits, inside that loop it will
>>>>> find more commands in the queue, and exeute them, so in the end it
>>>>> will only execute a bunch of "sleep(1000)".
>>>>>
>>>>> You're better of using a sleep in your module directly (not tell FS
>>>>> to sleep). Also you will need to poll the events in the loop
>>>>> somehow., checkout the perl sample here:
>>>>> http://wiki.freeswitch.org/**wiki/Esl#Simple_Perl_Example<http://wiki.freeswitch.org/wiki/Esl#Simple_Perl_Example>
>>>>>
>>>> Peter,
>>>>
>>>> I've changed to the below (adapted that Perl example) and I only hear
>>>> the prompt for digits when the webserver times out after 60 seconds.
>>>> Any time I make any call to ESL after the execute, I don't hear the
>>>> prompt in a timely fashion. This is so complicated as I'm not really
>>>> that good at software development, I'm more a systems/networks guy. I
>>>> feel like a total thickie here, but the docs don't seem to explain in
>>>> which order I should call things and expect to get replies, etc.
>>>>
>>>> $esl = new ESLconnection('localhost','**8021','ClueCon');
>>>>
>>>> $event = $esl->events('plain', 'ALL');
>>>> //$event = $esl->filter("Unique-ID",$**uuid);
>>>>
>>>> sleep (1);
>>>>
>>>> $res = $esl->execute("play_and_get_**digits","4 11 1 5000 #
>>>> ivr/ivr-please_enter_the_**phone_number.wav
>>>> ivr/ivr-that_was_an_invalid_**entry.wav target_num \d+", "$uuid");
>>>>
>>>> $job_uuid = $res->getHeader("Job-UUID");
>>>> $status = $res->getHeader("Reply-Text");
>>>>
>>>>
>>>>
>>>> //$foo = $uuid . ' target_num';
>>>>
>>>> echo $status;
>>>>
>>>> $stay_connected = 1;
>>>> while ( $stay_connected ) {
>>>>     $e = $esl->recvEventTimed(0);
>>>>     if ( $e ) {
>>>>         $ev_name = $e->getHeader("Event-Name");
>>>>         # Should we check for the $job_uuid to match the background job
>>>> ?
>>>>         //if ( $ev_name == 'BACKGROUND_JOB' ) {
>>>>         //    if ( $e->getHeader("Job-UUID") == $uuid ) {
>>>>         //        $call_result = $e->getBody();
>>>>         //        echo "Result of call to $target was $call_result\n\n";
>>>>         //    }
>>>>         //} else
>>>>         echo $ev_name;
>>>>         if ( $ev_name == 'EXECUTE_COMPLETE' ) {
>>>>             //my $digit = $e->getHeader("DTMF-Digit");
>>>>            // print "Received DTMF digit: $digit\n";
>>>>             //if ( $digit =~ m/\D/ ) {a
>>>>              //   print "Exiting...\n";
>>>>                 //if ( $e->getHeader("Job-UUID") == $job_uuid ) {
>>>>                 $stay_connected = 0;
>>>>                 //}
>>>>             //}
>>>>         } else {
>>>>       # Some other event
>>>>         }
>>>>     } else {
>>>>     # do other things while waiting for events
>>>>         //sleep (1);
>>>>     }
>>>> }
>>>>
>>>> $event=$esl->sendRecv("api uuid_getvar $uuid target_num");
>>>>
>>>>
>>>> $ret = $event->getBody();
>>>>
>>>> $esl->disconnect();
>>>>
>>>> Apologies for my newbishness and many thanks for your help so far.
>>>>
>>>> Hoping for an epiphany when I sudddenly understand what I am doing!
>>>>
>>>> Alex
>>>>
>>> Can anyone help with this? I updated to latest Git, thinking it may have
>>> been a bug, and still have the same problem, when I try to wait for
>>> events as in the above code, it stops the playback of the "enter the
>>> phone number" from happening until after the web server times out.
>>>
>>> Logically I can't see a reason for this, is it because you can't use an
>>> inbound connection activated by calling from curl in the dialplan? If
>>> this is indeed the case I could try rewriting this for outbound but I
>>> feel that will be more complex for a relative beginner programmer.
>>>
>>> If I have missed some vital or obvious concept feel free to chastise me
>>> :-).
>>>
>>> Thanks
>>>
>>> Alex
>>>
>>>
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From acrow at integrafin.co.uk  Fri Apr  6 23:51:03 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 06 Apr 2012 20:51:03 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
Message-ID: <4F7F4927.3090809@integrafin.co.uk>

On 06/04/12 20:34, Michael Collins wrote:
> Alex,
>
> I noticed that you are using the same thing as Michael Lutz from the 
> other thread we just discussed. Could you try recvEventTimed(1) 
> instead of recvEventTimed(0)? I'm curious to see if it changes the 
> behavior of your application at all.
>
> -MC
>
Michael,

I've tried 0, 1, 20, 200 and nothing makes a difference.

Cheers

Alex


From msc at freeswitch.org  Fri Apr  6 23:55:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 Apr 2012 12:55:02 -0700
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F7F4927.3090809@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
Message-ID: <CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>

Just to confirm... the symptom is that you start a play_and_get_digits
action and when you do the recvEventTime() function it stops the playback
of the pagd?

-MC

On Fri, Apr 6, 2012 at 12:51 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> On 06/04/12 20:34, Michael Collins wrote:
> > Alex,
> >
> > I noticed that you are using the same thing as Michael Lutz from the
> > other thread we just discussed. Could you try recvEventTimed(1)
> > instead of recvEventTimed(0)? I'm curious to see if it changes the
> > behavior of your application at all.
> >
> > -MC
> >
> Michael,
>
> I've tried 0, 1, 20, 200 and nothing makes a difference.
>
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From nestor at tiendalinux.com  Sat Apr  7 01:02:42 2012
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Fri, 06 Apr 2012 16:02:42 -0500
Subject: [Freeswitch-users] core dump on sip profile,
	fs version 2012-04-05
In-Reply-To: <CAKzWOxVe63=h4s6v-j_e_njnz_MFO7rfSb+54W_ddZyEHrYN5A@mail.gmail.com>
References: <4F7F4067.7000805@tiendalinux.com>
	<CAKzWOxVe63=h4s6v-j_e_njnz_MFO7rfSb+54W_ddZyEHrYN5A@mail.gmail.com>
Message-ID: <4F7F59F2.4050007@tiendalinux.com>

Done :)

On 04/06/2012 02:21 PM, Michael Collins wrote:
> This is great information! Of course, it's in the wrong place. :) 
> Please open a ticket at jira.freeswitch.org <http://jira.freeswitch.org>.
> -MC
>

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia

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From acrow at integrafin.co.uk  Sat Apr  7 01:20:55 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 06 Apr 2012 22:20:55 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
Message-ID: <4F7F5E37.7050308@integrafin.co.uk>

On 06/04/12 20:55, Michael Collins wrote:
> Just to confirm... the symptom is that you start a play_and_get_digits 
> action and when you do the recvEventTime() function it stops the 
> playback of the pagd?
>
> -MC
>
Yes,

In fact you never even hear the beginning of the pagd. You only hear it 
when the curl request from the dialplan call finally times out in the 
webserver (in my case nginx, 60 seconds, and FS logs the HTTP timeout 
the moment you hear the pagd prompt.)

Alex


From djbinter at gmail.com  Sat Apr  7 04:22:38 2012
From: djbinter at gmail.com (DJB International)
Date: Fri, 6 Apr 2012 17:22:38 -0700
Subject: [Freeswitch-users] BLF Presence
Message-ID: <CAAB3jWujuvEp=6k_=smUCO7OVp5nTDD8DdG5GYrrQ7rJbUgAZw@mail.gmail.com>

I currently have my Polycom 650 phone set up with presence enabled:

<feature feature.1.name="presence" feature.1.enabled="1" />


However, when I looked at the table sip_presence in sofia_reg_internal.db,
I do not see any information in there.

What I am trying to achieve is something similar to "sip show subscription"
to verify BLF subscriptions.

Is there something that I am missing here.  Please advise.

Thank you.
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From iruwen at gmx.net  Sat Apr  7 05:03:25 2012
From: iruwen at gmx.net (Iruwen)
Date: Sat, 07 Apr 2012 03:03:25 +0200
Subject: [Freeswitch-users] vm-keep-local-after-email vs vm-delete-file?
Message-ID: <4F7F925D.8080003@gmx.net>

Hi,

is vm-delete-file deprecated in favor of vm-keep-local-after-email?
I can only find the latter in the wiki now 
(http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-keep-local-after-email).
Are they supposed to do the same?
The recorded mp3 files aren't removed for me, anybody else experience that?

Regards


From anthony.minessale at gmail.com  Sat Apr  7 06:22:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Apr 2012 21:22:39 -0500
Subject: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
In-Reply-To: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>
References: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>
Message-ID: <CAKbxfG_YLLVZvxU-wF8qgnAx3AxQHo82ogn+-zH7CW=WreY7dw@mail.gmail.com>

You also must seperately enable odbc in the mod Sofia cfg.  Look for dsn in
all the configs.
On Apr 6, 2012 1:48 PM, "Yuriy Nasida" <nasida at live.ru> wrote:

>  Hello guys!
>
> I have set odbc mysql for internal profile. Can I remove
> sofia_reg_internal.db? I try to delete it but have lost my reg. After
> restart, FS creates this file again. What is core odbc support? FS just
> copy the data from file  sofia_reg_internal.db  to mysql via odbc ?
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jeff at jefflenk.com  Sat Apr  7 06:53:04 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 6 Apr 2012 19:53:04 -0700 (PDT)
Subject: [Freeswitch-users] vm-keep-local-after-email vs vm-delete-file?
In-Reply-To: <4F7F925D.8080003@gmx.net>
References: <4F7F925D.8080003@gmx.net>
Message-ID: <1333767184274-7444727.post@n2.nabble.com>

This looks like it may be backwards - please open a Jira on this.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vm-keep-local-after-email-vs-vm-delete-file-tp7444570p7444727.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From yehavi.bourvine at gmail.com  Sat Apr  7 10:39:26 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 7 Apr 2012 09:39:26 +0300
Subject: [Freeswitch-users] BLF Presence
In-Reply-To: <CAAB3jWujuvEp=6k_=smUCO7OVp5nTDD8DdG5GYrrQ7rJbUgAZw@mail.gmail.com>
References: <CAAB3jWujuvEp=6k_=smUCO7OVp5nTDD8DdG5GYrrQ7rJbUgAZw@mail.gmail.com>
Message-ID: <CAAtWKeEOgRtzyO+icNtizt2VtWZKi44k6nrNQx=KcK+itTz=vA@mail.gmail.com>

Hi,

  Are you asking that because presence does not work, or just because you
want to see the presence tables?

Note that turnning on the presence feature is not enough. You have to
define a speed dial ("buddy" in Polycom's jargon) to subscribe for
presence. You should have BW (buddy watch) enabled for each speed dial for
it to subscribe for presence.

                           Regards, __Yehavi:

2012/4/7 DJB International <djbinter at gmail.com>

> I currently have my Polycom 650 phone set up with presence enabled:
>
> <feature feature.1.name="presence" feature.1.enabled="1" />
>
>
> However, when I looked at the table sip_presence in sofia_reg_internal.db,
> I do not see any information in there.
>
> What I am trying to achieve is something similar to "sip show
> subscription" to verify BLF subscriptions.
>
> Is there something that I am missing here.  Please advise.
>
> Thank you.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From djbinter at gmail.com  Sat Apr  7 12:22:24 2012
From: djbinter at gmail.com (Dorn DJBinter)
Date: Sat, 7 Apr 2012 01:22:24 -0700
Subject: [Freeswitch-users] BLF Presence
In-Reply-To: <CAAtWKeEOgRtzyO+icNtizt2VtWZKi44k6nrNQx=KcK+itTz=vA@mail.gmail.com>
References: <CAAB3jWujuvEp=6k_=smUCO7OVp5nTDD8DdG5GYrrQ7rJbUgAZw@mail.gmail.com>
	<CAAtWKeEOgRtzyO+icNtizt2VtWZKi44k6nrNQx=KcK+itTz=vA@mail.gmail.com>
Message-ID: <-159227813343798664@unknownmsgid>

Thank you, Yehavi.

Presence is working fine.  I just want to see the status in sip_presence
table which I assume that it would have similar info as "sip show
subscriptions", but on my FreeSWITCH system, that table does not populate
any info at all which I wonder whether I need to adjust any variables
somewhere.



On Apr 6, 2012, at 11:41 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

Hi,

  Are you asking that because presence does not work, or just because you
want to see the presence tables?

Note that turnning on the presence feature is not enough. You have to
define a speed dial ("buddy" in Polycom's jargon) to subscribe for
presence. You should have BW (buddy watch) enabled for each speed dial for
it to subscribe for presence.

                           Regards, __Yehavi:

2012/4/7 DJB International <djbinter at gmail.com>

> I currently have my Polycom 650 phone set up with presence enabled:
>
> <feature feature.1.name="presence" feature.1.enabled="1" />
>
>
> However, when I looked at the table sip_presence in sofia_reg_internal.db,
> I do not see any information in there.
>
> What I am trying to achieve is something similar to "sip show
> subscription" to verify BLF subscriptions.
>
> Is there something that I am missing here.  Please advise.
>
> Thank you.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From wesleyakio at tuntscorp.com  Sat Apr  7 13:33:19 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Sat, 7 Apr 2012 06:33:19 -0300
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F7F5E37.7050308@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
Message-ID: <CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>

Maybe the dialplan is waiting for curl to return... Can you see that at the
logs?
Em 06/04/2012 18:21, "Alex Crow" <acrow at integrafin.co.uk> escreveu:

> On 06/04/12 20:55, Michael Collins wrote:
> > Just to confirm... the symptom is that you start a play_and_get_digits
> > action and when you do the recvEventTime() function it stops the
> > playback of the pagd?
> >
> > -MC
> >
> Yes,
>
> In fact you never even hear the beginning of the pagd. You only hear it
> when the curl request from the dialplan call finally times out in the
> webserver (in my case nginx, 60 seconds, and FS logs the HTTP timeout
> the moment you hear the pagd prompt.)
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From amilkhanzada at gmail.com  Sat Apr  7 21:40:20 2012
From: amilkhanzada at gmail.com (LearningFS)
Date: Sat, 7 Apr 2012 10:40:20 -0700 (PDT)
Subject: [Freeswitch-users] Using Softphone on same machine as FreeSwitch?
Message-ID: <33649239.post@talk.nabble.com>


Hi guys,

I am on Ubuntu 10.04 inside a VirtualBox virtual machine (my host OS is
Windows 7).
I am running FreeSwitch on Ubuntu.

However, when I try to connect with any softphone within my Ubuntu, it
cannot connect to FreeSwitch and FS does not receive the connection requests
on command prompt.
I tried connecting to port 5060 on the "inet addr:" shown in ifconfig.

I actually was able to get an android phone to connect to my windows public
IP address after setting up port forwarding, but is it possible for me to
get an Ubuntu softphone to connect as well so I can keep all my testing in
one machine?

Also, I want to test out mod_sms. I have X-Lite on my Windows machine, I
guess this is sufficient.

Thanks and sorry if this was answered before. I tried searching for it.
-- 
View this message in context: http://old.nabble.com/Using-Softphone-on-same-machine-as-FreeSwitch--tp33649239p33649239.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From david at harktech.com  Sat Apr  7 19:35:16 2012
From: david at harktech.com (David George)
Date: Sat, 07 Apr 2012 11:35:16 -0400
Subject: [Freeswitch-users] Need help with FreeTDM config
Message-ID: <4F805EB4.70509@harktech.com>

I am writing an application that uses FreeTDM with a Digium TE-220 Dual 
T1 card.  I started by looking at the testanalog sample and found that I 
also needed to add ftdm_global_configuration after ftdm_global_init.  I 
am now able to place a call, but I get a segfault early on referencing 
sigmsg->channel.

I am using an Adtran TSU600e for a channel bank and have a mixture of 
FXS and FXO cards in it.  Right now I am just testing with a phone 
plugged into an FXS port configured for TANDEM_LS.  The T1 is configured 
AMI/D4 for channelized T1 with robbed bit signalling.  The T1 is up and 
I have no alarms.

There isn't a lot of information on the FreeSWITCH Wiki or on the 
Sangoma FreeTDM Wiki for T1 RBS.  There is one mention that I should use 
channel type of fxo-channel or fxs-channel for RBS ( 
http://wiki.freeswitch.org/wiki/FreeTDM ).  I saw another place that 
says to use em-channel, but it may be specific to running Dahdi on 
Sangoma cards ( http://wiki.sangoma.com/FreeSWITCH-Dahdi-Mode ).

I have tried both configurations, but have gotten further with the 
fxo-channel config than with the em-channel config.

With the channel set to fxo-channel I get a segfault when I place a call 
and the first sigmsg->channel is referenced.

I can open a Jira and add backtraces, but I wanted to check if I 
probably had a configuration issue first.

E&M configuration I tried:
[span zt tsu600e]
trunk_type => EM
em-channel => 1-4
;fxo-channel => 1-4
;fxs-channel => 5-6
;fxo-channel => 9-12

FXO config I tried:
[span zt tsu600e]
fxo-channel => 1-4

Channel 1-4 is really an FXS card.  I am using the reverse like I used 
to with Asterisk.  Not sure if it is correct or not.

Thanks,
David

-- 
David George
Hark Technologies



From millyman12721 at gmail.com  Sat Apr  7 21:28:39 2012
From: millyman12721 at gmail.com (millerman33)
Date: Sat, 7 Apr 2012 10:28:39 -0700 (PDT)
Subject: [Freeswitch-users] skypopen mod wont let me call out
Message-ID: <1333819719037-7446255.post@n2.nabble.com>

Every time I place a call in skype through freeswitch I keep getting this
error message

freeswitch at Home-PC>2012-04-07 13:11.26.261968 [ERR] mod_skyopen.c:2330
    [git-]  [ERRORA     2330  ][interface1      ][RINGING,RINGING] No
session_uuid_str??? 


Thats every outbound call, im a newbie to this maybe I installed something
wrong. 




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-mod-wont-let-me-call-out-tp7446255p7446255.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From acrow at integrafin.co.uk  Sun Apr  8 00:38:10 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sat, 07 Apr 2012 21:38:10 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
Message-ID: <4F80A5B2.7010703@integrafin.co.uk>

On 07/04/12 10:33, Wesley Akio wrote:
>
> Maybe the dialplan is waiting for curl to return... Can you see that 
> at the logs?
>
>

Thanks Wesley,

But the curl should return as soon as the event is detected, and surely 
the queued request for PAGD should proceed regardless of my polling for 
events after it.  I've even tried listening for all events and logging 
them but I see nothing in the logs and still have to wait for the web 
server to timeout. The really odd thing is that if I don't do anything 
else in my PHP after the PAGD ESL call then I hear the message playing, 
as soon as I try to do anything else (including polling for events) I don't.

I am just curious as to a) if I am trying to do something unsupported or 
b) I have found a bug.

Cheers

Alex





From gcd at i.ph  Sun Apr  8 01:56:35 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Sun, 8 Apr 2012 05:56:35 +0800
Subject: [Freeswitch-users] Using Softphone on same machine as
	FreeSwitch?
In-Reply-To: <33649239.post@talk.nabble.com>
References: <33649239.post@talk.nabble.com>
Message-ID: <CAHKqMb635kBHY4-y-bh=4kYSL7xYmAHq8jqiJBkktwMCAMED7Q@mail.gmail.com>

the x-lite is okay but IMHO add at least 2 real phones (IP or ATA) for
complete testing e.g. conference calls, call forwarding, etc.

On Sun, Apr 8, 2012 at 1:40 AM, LearningFS <amilkhanzada at gmail.com> wrote:

>
> Hi guys,
>
> I am on Ubuntu 10.04 inside a VirtualBox virtual machine (my host OS is
> Windows 7).
> I am running FreeSwitch on Ubuntu.
>
> However, when I try to connect with any softphone within my Ubuntu, it
> cannot connect to FreeSwitch and FS does not receive the connection
> requests
> on command prompt.
> I tried connecting to port 5060 on the "inet addr:" shown in ifconfig.
>
> I actually was able to get an android phone to connect to my windows public
> IP address after setting up port forwarding, but is it possible for me to
> get an Ubuntu softphone to connect as well so I can keep all my testing in
> one machine?
>
> Also, I want to test out mod_sms. I have X-Lite on my Windows machine, I
> guess this is sufficient.
>
> Thanks and sorry if this was answered before. I tried searching for it.
> --
> View this message in context:
> http://old.nabble.com/Using-Softphone-on-same-machine-as-FreeSwitch--tp33649239p33649239.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From manjiri05_deshpande at yahoo.co.in  Sun Apr  8 12:21:34 2012
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Sun, 8 Apr 2012 16:21:34 +0800 (SGT)
Subject: [Freeswitch-users] Fwd:
Message-ID: <1333873294.39986.BPMail_high_noncarrier@web192504.mail.sg3.yahoo.com>


http://www.caliearthquakekits.com/entrance.php?jkypilo=237&xuhutohu=85


From gmaruzz at gmail.com  Sun Apr  8 02:13:32 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 8 Apr 2012 00:13:32 +0200
Subject: [Freeswitch-users] skypopen mod wont let me call out
In-Reply-To: <1333819719037-7446255.post@n2.nabble.com>
References: <1333819719037-7446255.post@n2.nabble.com>
Message-ID: <CALXCt0p6j1zWSCsR8aj=7PfHgntYfD=LcXNnfrOAx3Bw+-r3Fw@mail.gmail.com>

please,
follow literally, step by step, the wiki page
http://wiki.freeswitch.org/skypopen
Those are cut and paste instructions, very much reliables.

Then, if you still have problem, follow the instructions on how to
report a problem.

Let us know how it goes.

-giovanni

On 4/7/12, millerman33 <millyman12721 at gmail.com> wrote:
> Every time I place a call in skype through freeswitch I keep getting this
> error message
>
> freeswitch at Home-PC>2012-04-07 13:11.26.261968 [ERR] mod_skyopen.c:2330
>     [git-]  [ERRORA     2330  ][interface1      ][RINGING,RINGING] No
> session_uuid_str???
>
>
> Thats every outbound call, im a newbie to this maybe I installed something
> wrong.
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/skypopen-mod-wont-let-me-call-out-tp7446255p7446255.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From maximsuzuki at gmail.com  Sun Apr  8 06:52:34 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Sat, 7 Apr 2012 22:52:34 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
Message-ID: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>

Wondering how to generate client certificates on Windows?  Is there a
windows equivalent of:

 

./gentls_cert create_client -cn Client1 -out Client1

 

I am using the latest source from Git.

 

Thanks.

 

 

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From peter.olsson at visionutveckling.se  Sun Apr  8 16:13:35 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 8 Apr 2012 12:13:35 +0000
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>

Maybe this is what you're lokking for?

http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_%28win32%29

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Maxim Suzuki [maximsuzuki at gmail.com]
Skickat: den 8 april 2012 04:52
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] generating TLS client certificates on Windows

Wondering how to generate client certificates on Windows?  Is there a windows equivalent of:

./gentls_cert create_client -cn Client1 -out Client1

I am using the latest source from Git.

Thanks.


!DSPAM:4f817daf32762008726396!


From millyman12721 at gmail.com  Sun Apr  8 21:22:09 2012
From: millyman12721 at gmail.com (millerman33)
Date: Sun, 8 Apr 2012 10:22:09 -0700 (PDT)
Subject: [Freeswitch-users] skypopen mod wont let me call out
In-Reply-To: <CALXCt0p6j1zWSCsR8aj=7PfHgntYfD=LcXNnfrOAx3Bw+-r3Fw@mail.gmail.com>
References: <1333819719037-7446255.post@n2.nabble.com>
	<CALXCt0p6j1zWSCsR8aj=7PfHgntYfD=LcXNnfrOAx3Bw+-r3Fw@mail.gmail.com>
Message-ID: <1333905729799-7447994.post@n2.nabble.com>

Thank you Giovanni for your response. I have downloaded and build this
program about four times now. I ran the freeswitch console in debug mode and
it showed this when I attempted to make a call.


04-08 13:14:05.983464 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface1     ][DIALING,UNPLACD] READING: |||CALL 85261 STATUS R

04-08 13:14:05.983464 [DEBUG] skypopen_protocol.c:713     [git-] [DEBUG
13  ][interface1     ][DIALING,ROUTING] skype_call: 85261 is now ROUTIN
04-08 13:14:05.983464 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface2     ][DIALING,UNPLACD] READING: |||CALL 85261 STATUS R

04-08 13:14:05.983464 [DEBUG] skypopen_protocol.c:713     [git-] [DEBUG
13  ][interface2     ][DIALING,ROUTING] skype_call: 85261 is now ROUTIN
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface1     ][DIALING,ROUTING] READING: |||CALL 85261 STATUS R

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:616     [git-] [DEBUG
16  ][interface1     ][RINGING,RINGING] Our remote party in skype_call
RINGING
04-08 13:14:12.308826 [ERR] mod_skypopen.c:2330          [git-] [ERRORA
30 ][interface1     ][RINGING,RINGING] No session_uuid_str???
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:620     [git-] [DEBUG
20  ][interface1     ][RINGING,RINGING] We are getting the RINGING from
e probably canceled, trying to get out hanging up call id: 85261.
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:1304    [git-] [DEBUG
304 ][interface1     ][RINGING,RINGING] SENDING: |||ALTER CALL 85261 EN
|||
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface2     ][DIALING,ROUTING] READING: |||CALL 85261 STATUS R

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:616     [git-] [DEBUG
16  ][interface2     ][RINGING,RINGING] Our remote party in skype_call
RINGING
04-08 13:14:12.308826 [ERR] mod_skypopen.c:2330          [git-] [ERRORA
30 ][interface2     ][RINGING,RINGING] No session_uuid_str???
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:620     [git-] [DEBUG
20  ][interface2     ][RINGING,RINGING] We are getting the RINGING from
e probably canceled, trying to get out hanging up call id: 85261.
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:1304    [git-] [DEBUG
304 ][interface2     ][RINGING,RINGING] SENDING: |||ALTER CALL 85261 EN
|||
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:1304    [git-] [DEBUG
304 ][interface1     ][RINGING,RINGING] SENDING: |||ALTER CALL 85261 HA

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:1304    [git-] [DEBUG
304 ][interface2     ][RINGING,RINGING] SENDING: |||ALTER CALL 85261 HA

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:630     [git-] [DEBUG
30  ][interface1     ][DOWN,IDLE] we're now DOWN
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1420         [git-] [DEBUG
420 ][interface1     ][DOWN,IDLE] skype call ended
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1440         [git-] [DEBUG
440 ][interface1     ][DOWN,IDLE] no session
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1443         [git-] [DEBUG
443 ][interface1     ][DOWN,IDLE] audio tcp threads to DIE
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1453         [git-] [DEBUG
453 ][interface1     ][DOWN,IDLE] audio tcp srv thread DEAD 0
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1463         [git-] [DEBUG
463 ][interface1     ][DOWN,IDLE] audio tcp cli thread DEAD 0
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface1     ][IDLE,IDLE] READING: |||ALTER CALL 85261 END HANG

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:630     [git-] [DEBUG
30  ][interface2     ][DOWN,IDLE] we're now DOWN
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1420         [git-] [DEBUG
420 ][interface2     ][DOWN,IDLE] skype call ended
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1440         [git-] [DEBUG
440 ][interface2     ][DOWN,IDLE] no session
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1443         [git-] [DEBUG
443 ][interface2     ][DOWN,IDLE] audio tcp threads to DIE
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1453         [git-] [DEBUG
453 ][interface2     ][DOWN,IDLE] audio tcp srv thread DEAD 0
04-08 13:14:12.308826 [DEBUG] mod_skypopen.c:1463         [git-] [DEBUG
463 ][interface2     ][DOWN,IDLE] audio tcp cli thread DEAD 0
04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface2     ][IDLE,IDLE] READING: |||ALTER CALL 85261 END HANG

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface1     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action fa

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:228     [git-] [DEBUG
28  ][interface1     ][IDLE,IDLE] Skype got ERROR about a failed action
y TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action faile

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface2     ][IDLE,IDLE] READING: |||ERROR 559 CALL: Action fa

04-08 13:14:12.308826 [DEBUG] skypopen_protocol.c:228     [git-] [DEBUG
28  ][interface2     ][IDLE,IDLE] Skype got ERROR about a failed action
y TRYING to HANGUP A CALL), no problem: |||ERROR 559 CALL: Action faile

04-08 13:14:12.388831 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface1     ][IDLE,IDLE] READING: |||CALL 85261 STATUS CANCELL

04-08 13:14:12.388831 [DEBUG] skypopen_protocol.c:677     [git-] [DEBUG
77  ][interface1     ][IDLE,CANCLED] we tried to call Skype on skype_ca
and Skype has now CANCELLED
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1420         [git-] [DEBUG
420 ][interface1     ][DOWN,CANCLED] skype call ended
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1440         [git-] [DEBUG
440 ][interface1     ][DOWN,CANCLED] no session
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1443         [git-] [DEBUG
443 ][interface1     ][DOWN,CANCLED] audio tcp threads to DIE
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1453         [git-] [DEBUG
453 ][interface1     ][DOWN,CANCLED] audio tcp srv thread DEAD 0
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1463         [git-] [DEBUG
463 ][interface1     ][DOWN,CANCLED] audio tcp cli thread DEAD 0
04-08 13:14:12.388831 [DEBUG] skypopen_protocol.c:173     [git-] [DEBUG
73  ][interface2     ][IDLE,IDLE] READING: |||CALL 85261 STATUS CANCELL

04-08 13:14:12.388831 [DEBUG] skypopen_protocol.c:677     [git-] [DEBUG
77  ][interface2     ][IDLE,CANCLED] we tried to call Skype on skype_ca
and Skype has now CANCELLED
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1420         [git-] [DEBUG
420 ][interface2     ][DOWN,CANCLED] skype call ended
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1440         [git-] [DEBUG
440 ][interface2     ][DOWN,CANCLED] no session
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1443         [git-] [DEBUG
443 ][interface2     ][DOWN,CANCLED] audio tcp threads to DIE
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1453         [git-] [DEBUG
453 ][interface2     ][DOWN,CANCLED] audio tcp srv thread DEAD 0
04-08 13:14:12.388831 [DEBUG] mod_skypopen.c:1463         [git-] [DEBUG
463 ][interface2     ][DOWN,CANCLED] audio tcp cli thread DEAD 0 

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-mod-wont-let-me-call-out-tp7446255p7447994.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From maximsuzuki at gmail.com  Sun Apr  8 16:25:32 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Sun, 8 Apr 2012 08:25:32 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
Message-ID: <000001cd1582$b2713db0$1753b910$@gmail.com>

Thanks but this wiki entry only describes how to generate the CA root and
server certificates on Windows:

cafile.pem
agent.pem


I'm looking for the equivalent of the following command on Windows:

./gentls_cert create_client -cn Client1 -out Client1 

... to generate: 

Client1.pem

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: Sunday, April 08, 2012 8:14 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] generating TLS client certificates on
Windows

Maybe this is what you're lokking for?

http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_%28win32%29

/Peter
________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f?r Maxim Suzuki
[maximsuzuki at gmail.com]
Skickat: den 8 april 2012 04:52
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] generating TLS client certificates on Windows

Wondering how to generate client certificates on Windows?  Is there a
windows equivalent of:

./gentls_cert create_client -cn Client1 -out Client1

I am using the latest source from Git.

Thanks.


!DSPAM:4f817daf32762008726396!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From amilkhanzada at gmail.com  Mon Apr  9 00:18:51 2012
From: amilkhanzada at gmail.com (LearningFS)
Date: Sun, 8 Apr 2012 13:18:51 -0700 (PDT)
Subject: [Freeswitch-users] Using Softphone on same machine as FreeSwitch?
Message-ID: <33649239.post@talk.nabble.com>


Hi guys,

I am on Ubuntu 10.04 inside a VirtualBox virtual machine (my host OS is
Windows 7).
I am running FreeSwitch on Ubuntu.

However, when I try to connect with any softphone within my Ubuntu, it
cannot connect to FreeSwitch and FS does not receive the connection requests
on command prompt.
I tried connecting to port 5060 on the "inet addr:" shown in ifconfig. I
double-checked the ports in my FS conf directory.

I don't think it will work for me to use the public IP address of my Ubuntu
because this is all hidden behind my virtual machine NAT + my router +
symantec and windows firewall on my Windows machine. I tried doing a bunch
of port forwarding stuff, but this did not work out very well...

Is there any way to connect a softphone to FS internally within Ubuntu?

Also, I want to test out mod_sms. I think qutecom is suitable for this
right?

Thanks and sorry if this was answered before. I tried searching for it.
-- 
View this message in context: http://old.nabble.com/Using-Softphone-on-same-machine-as-FreeSwitch--tp33649239p33649239.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From wesleyakio at tuntscorp.com  Mon Apr  9 04:08:32 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Sun, 8 Apr 2012 21:08:32 -0300
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F80A5B2.7010703@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
	<4F80A5B2.7010703@integrafin.co.uk>
Message-ID: <CAGTsDSvSOq=F6VGsZPZy2bzD+uVCDcA+6RmDsQnr284a-YU60Q@mail.gmail.com>

The script that does the polling is the one called by curl? Remember curl
will return only when that script finishes.
Em 07/04/2012 17:38, "Alex Crow" <acrow at integrafin.co.uk> escreveu:

> On 07/04/12 10:33, Wesley Akio wrote:
> >
> > Maybe the dialplan is waiting for curl to return... Can you see that
> > at the logs?
> >
> >
>
> Thanks Wesley,
>
> But the curl should return as soon as the event is detected, and surely
> the queued request for PAGD should proceed regardless of my polling for
> events after it.  I've even tried listening for all events and logging
> them but I see nothing in the logs and still have to wait for the web
> server to timeout. The really odd thing is that if I don't do anything
> else in my PHP after the PAGD ESL call then I hear the message playing,
> as soon as I try to do anything else (including polling for events) I
> don't.
>
> I am just curious as to a) if I am trying to do something unsupported or
> b) I have found a bug.
>
> Cheers
>
> Alex
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Mon Apr  9 07:03:06 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 8 Apr 2012 20:03:06 -0700
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F80A5B2.7010703@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
	<4F80A5B2.7010703@integrafin.co.uk>
Message-ID: <CAKzWOxVnt37gbBdyQF=TowjfuL+nHzvRbGoudTqYcriDJhsR6w@mail.gmail.com>

On Sat, Apr 7, 2012 at 1:38 PM, Alex Crow <acrow at integrafin.co.uk> wrote:

> On 07/04/12 10:33, Wesley Akio wrote:
> >
> > Maybe the dialplan is waiting for curl to return... Can you see that
> > at the logs?
> >
> >
>
> Thanks Wesley,
>
> But the curl should return as soon as the event is detected, and surely
> the queued request for PAGD should proceed regardless of my polling for
> events after it.  I've even tried listening for all events and logging
> them but I see nothing in the logs and still have to wait for the web
> server to timeout. The really odd thing is that if I don't do anything
> else in my PHP after the PAGD ESL call then I hear the message playing,
> as soon as I try to do anything else (including polling for events) I
> don't.
>
> I am just curious as to a) if I am trying to do something unsupported or
> b) I have found a bug.
>
>
I suspect there are better tools at your disposal for doing what you are
trying to do. If I were in your shoes (someone who obviously has Web dev
and PHP experience) then I would cut my losses with method and head
straight for HTTAPI. My guess is that the initial learning curve will be a
bit steep but that once you get over the hump you will be building awesome
applications without all the weird issues.

-MC
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From mitch.capper at gmail.com  Mon Apr  9 07:03:01 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 8 Apr 2012 20:03:01 -0700
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <000001cd1582$b2713db0$1753b910$@gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
Message-ID: <CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>

Hi Maxim,
Take a look at the gentls_cert script it has the openssl.cfg and
commands used to generate the certs. As the wiki details getting
openssl working in windows you should be able to replicate it without
issue.

~mitch


From maximsuzuki at gmail.com  Mon Apr  9 07:20:51 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Sun, 8 Apr 2012 23:20:51 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
	<CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
Message-ID: <CAPwVh_hefmF+2QVRw=e1srty8CwPESCaO0M3zREHL=P08qtOcA@mail.gmail.com>

Thanks Mitch.  That looks like the best approch.
On Apr 8, 2012 11:04 PM, "Mitch Capper" <mitch.capper at gmail.com> wrote:

> Hi Maxim,
> Take a look at the gentls_cert script it has the openssl.cfg and
> commands used to generate the certs. As the wiki details getting
> openssl working in windows you should be able to replicate it without
> issue.
>
> ~mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From dinhthuc1303 at gmail.com  Mon Apr  9 07:22:25 2012
From: dinhthuc1303 at gmail.com (dinh thuc)
Date: Mon, 9 Apr 2012 10:22:25 +0700
Subject: [Freeswitch-users] tone detect
Message-ID: <CAKMOEZ=KmQo0ebXCRgb4uC5qV_hsBPOP9R89Rjh5eAhcv57CTA@mail.gmail.com>

Hi

I want to test tone detect in Freeswitch as: detect tone and events

I want to create virtual: send and get tone depend on events

I have read http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detectand
http://wiki.freeswitch.org/wiki/Mod_spandsp#start_tone_detect

Do you know the way to establish freeswitch to get events tone decect ?
(that mean is vitual tone to test in freeswitch)

B& R
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From acrow at integrafin.co.uk  Mon Apr  9 12:21:03 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 09 Apr 2012 09:21:03 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAGTsDSvSOq=F6VGsZPZy2bzD+uVCDcA+6RmDsQnr284a-YU60Q@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
	<4F80A5B2.7010703@integrafin.co.uk>
	<CAGTsDSvSOq=F6VGsZPZy2bzD+uVCDcA+6RmDsQnr284a-YU60Q@mail.gmail.com>
Message-ID: <4F829BEF.9020605@integrafin.co.uk>

On 09/04/12 01:08, Wesley Akio wrote:
>
> The script that does the polling is the one called by curl? Remember 
> curl will return only when that script finishes.
>
>

Wesley,

Yes. The script sends an ESL execute for the pagd and tries to poll for 
a response. Are you saying that I won't hear the PAGD until the script 
returns, so waiting for a response actually prevents it from playing 
until the script times out?

Cheers

Alex


From acrow at integrafin.co.uk  Mon Apr  9 12:21:59 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 09 Apr 2012 09:21:59 +0100
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <CAKzWOxVnt37gbBdyQF=TowjfuL+nHzvRbGoudTqYcriDJhsR6w@mail.gmail.com>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
	<4F80A5B2.7010703@integrafin.co.uk>
	<CAKzWOxVnt37gbBdyQF=TowjfuL+nHzvRbGoudTqYcriDJhsR6w@mail.gmail.com>
Message-ID: <4F829C27.1070605@integrafin.co.uk>


>
> I suspect there are better tools at your disposal for doing what you 
> are trying to do. If I were in your shoes (someone who obviously has 
> Web dev and PHP experience) then I would cut my losses with method and 
> head straight for HTTAPI. My guess is that the initial learning curve 
> will be a bit steep but that once you get over the hump you will be 
> building awesome applications without all the weird issues.
>
> -MC

Thanks Michael, I will have a look at it. It does say on the wiki it 
needs a cgi, but presumable a PHP page will work just as well?

Cheers

Alex



From wesleyakio at tuntscorp.com  Mon Apr  9 14:19:03 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Mon, 9 Apr 2012 07:19:03 -0300
Subject: [Freeswitch-users] Dialplan/PHP esl variable problem.
In-Reply-To: <4F829BEF.9020605@integrafin.co.uk>
References: <4F74771E.5050808@integrafin.co.uk>
	<1A52405B-F63B-4FE7-BDAF-2AD154593485@visionutveckling.se>
	<4F781477.2020602@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084DF1@cantor.std.visionutv.se>
	<4F7824ED.7000807@integrafin.co.uk>
	<1FFF97C269757C458224B7C895F35F15084E12@cantor.std.visionutv.se>
	<4F78827F.7020401@integrafin.co.uk>
	<4F7F2861.1030007@integrafin.co.uk>
	<CAKbxfG_yFHzEYOWKmN+rxQzuw3xJj2j-mdcUUhAGC+=7LhR+GQ@mail.gmail.com>
	<4F7F4248.3040404@integrafin.co.uk>
	<CAKzWOxV2AKo34isjrk6JXJFA+AWP0_dQf=+ryW7k-zr42Cpu0A@mail.gmail.com>
	<4F7F4927.3090809@integrafin.co.uk>
	<CAKzWOxW7dYgCArGPRcYqaVC6rgAmqmwLfkqaQWb9gA5+unc-Vw@mail.gmail.com>
	<4F7F5E37.7050308@integrafin.co.uk>
	<CAGTsDSuhY=5SqmP9OXWxdr3HRk9AKmC+RFFijBZbzCkKkE6k6w@mail.gmail.com>
	<4F80A5B2.7010703@integrafin.co.uk>
	<CAGTsDSvSOq=F6VGsZPZy2bzD+uVCDcA+6RmDsQnr284a-YU60Q@mail.gmail.com>
	<4F829BEF.9020605@integrafin.co.uk>
Message-ID: <CAGTsDSvrRyrwCp5a1T7rH+KfBMs_xFGMvX1O7xpZCtxwZXgWFQ@mail.gmail.com>

I'll try it today to be certain but yes, I'm assuming your dialplan
execution blocks on curl since only one application will be run on the call
at a time...

One thing you can try is to schedule curl to run on the background and park
the call right after, that way you will be certain curl is not the issue
since it will be running on another thread...

I would try outbound sockets though. They seem more fit to your scenario...

Em 09/04/2012 05:22, "Alex Crow" <acrow at integrafin.co.uk> escreveu:
>
> On 09/04/12 01:08, Wesley Akio wrote:
> >
> > The script that does the polling is the one called by curl? Remember
> > curl will return only when that script finishes.
> >
> >
>
> Wesley,
>
> Yes. The script sends an ESL execute for the pagd and tries to poll for
> a response. Are you saying that I won't hear the PAGD until the script
> returns, so waiting for a response actually prevents it from playing
> until the script times out?
>
> Cheers
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From dujinfang at gmail.com  Mon Apr  9 16:48:05 2012
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 9 Apr 2012 20:48:05 +0800
Subject: [Freeswitch-users] Open Bugs on Jira and Call for help
In-Reply-To: <CB98AD98.308CC%krice@freeswitch.org>
References: <CB98AD98.308CC%krice@freeswitch.org>
Message-ID: <77488073CB0942C08D06A35B46EE6D63@gmail.com>

Thanks Ken.

see http://jira.freeswitch.org/browse/FS-4088

questions:

1) I see confs both in mod_xxx/conf and conf/vanilla, should I modify both? I might miss the vanilla story, would anyone tell it briefly?

2) wiki saids http://wiki.freeswitch.org/wiki/Mod_conference

conference-flags Can be any combination of wait-mod and waste-bandwidth. wait-mod  

but I only see  

for (i = 0; i < argc && argv[i]; i++) {
if (!strcasecmp(argv[i], "wait-mod")) {
*f |= CFLAG_WAIT_MOD;
} else if (!strcasecmp(argv[i], "video-floor-only")) {
*f |= CFLAG_VID_FLOOR;
} else if (!strcasecmp(argv[i], "video-bridge")) {
*f |= CFLAG_VIDEO_BRIDGE;
}


is wiki wrong or should we patch mod_conference.c to enable CFLAG_WASTE_FLAG ?  I see this flag is not actually used.

7.  

On Thursday, March 29, 2012 at 12:59 AM, Ken Rice wrote:

> Re: [Freeswitch-users] Open Bugs on Jira and Call for help Actually we can opena  tracking bug on Jira or just email me the diffs I can review and commit that way...
>  
> Chjeck out git-format-patch its a good way to format a patch so you get credit also
>  
> K
>  
>  
> On 3/28/12 11:41 AM, "Seven Du" <dujinfang at gmail.com> wrote:
>  
> >  
> > Cool. I'll update some of my open bugs on jira. And for documentation help, I suggest also address the indents, especially don't mix spaces and tabs, I guess all like tabs? Also, is there any coordination of which conf files who's working on or should all choose randomly? And show we commit to git or report on jira?
> >  
> >    
> >  
> > On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote:
> >  
> > >  
> > > Open Bugs on Jira and Call for help If you have open bugs on Jira, please make sure they are up to date on any all information....  
> > >  
> > > If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We?re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible...
> > >  
> > > Also, I?m looking for some configuration documentation help. If you?ve pulled git in the past few days you?ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are.  The end goal is to get all the options documented in their default states.  
> > > What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs...  
> > >  
> > > K  
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >  
> > > 
> > > 
> > >  
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >  
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >       
> >   
> >   
> >   
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >  
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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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From krice at freeswitch.org  Mon Apr  9 17:01:30 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 09 Apr 2012 08:01:30 -0500
Subject: [Freeswitch-users] Open Bugs on Jira and Call for help
In-Reply-To: <77488073CB0942C08D06A35B46EE6D63@gmail.com>
Message-ID: <CBA847DA.30E76%krice@freeswitch.org>

The stuff in mod_XXX/conf should be exhaustive for the module...

The stuff in conf/${style}/* should be a working config specific to the
example being presented in ${style}


As far as wiki differening from code, I think we should go with what the
code says...   There are a few cases where this might not be true (in the
even of say a regression)

K


On 4/9/12 7:48 AM, "Seven Du" <dujinfang at gmail.com> wrote:

>  
> Thanks Ken.
> 
> see http://jira.freeswitch.org/browse/FS-4088
> 
> questions:
> 
> 1) I see confs both in mod_xxx/conf and conf/vanilla, should I modify both? I
> might miss the vanilla story, would anyone tell it briefly?
> 
> 2) wiki saids http://wiki.freeswitch.org/wiki/Mod_conference
> 
> conference-flags Can be any combination of wait-mod and waste-bandwidth.
> wait-mod 
> 
> but I only see 
> 
> for (i = 0; i < argc && argv[i]; i++) {
> if (!strcasecmp(argv[i], "wait-mod")) {
> *f |= CFLAG_WAIT_MOD;
> } else if (!strcasecmp(argv[i], "video-floor-only")) {
> *f |= CFLAG_VID_FLOOR;
> } else if (!strcasecmp(argv[i], "video-bridge")) {
> *f |= CFLAG_VIDEO_BRIDGE;
> }
> 
> is wiki wrong or should we patch mod_conference.c to enable CFLAG_WASTE_FLAG ?
> I see this flag is not actually used.
> 
> 7.
>   
> 
> On Thursday, March 29, 2012 at 12:59 AM, Ken Rice wrote:
>  
>>  
>> Re: [Freeswitch-users] Open Bugs on Jira and Call for help Actually we can
>> opena  tracking bug on Jira or just email me the diffs I can review and
>> commit that way...
>> 
>> Chjeck out git-format-patch its a good way to format a patch so you get
>> credit also
>> 
>> K
>> 
>> 
>> On 3/28/12 11:41 AM, "Seven Du" <dujinfang at gmail.com> wrote:
>> 
>>>  
>>> Cool. I'll update some of my open bugs on jira. And for documentation help,
>>> I suggest also address the indents, especially don't mix spaces and tabs, I
>>> guess all like tabs? Also, is there any coordination of which conf files
>>> who's working on or should all choose randomly? And show we commit to git or
>>> report on jira?
>>> 
>>>   
>>> 
>>> On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote:
>>>  
>>>>  
>>>> Open Bugs on Jira and Call for help If you have open bugs on Jira, please
>>>> make sure they are up to date on any all information....
>>>> 
>>>> If you wish to address a specific bug, please visit us on the weekly
>>>> conference call so we can address them... We?re getting close to time for
>>>> rolling Release Candidate for 1.2 and we want to address as many of these
>>>> as possible...
>>>> 
>>>> Also, I?m looking for some configuration documentation help. If you?ve
>>>> pulled git in the past few days you?ll notice that in the module src
>>>> directories there is now a conf directory, we need some help with making
>>>> sure all the configuration options are documented and what their default
>>>> settings are.  The end goal is to get all the options documented in their
>>>> default states.
>>>> What skills do you need? The ability to read a little C, and the ability to
>>>> use a text editor to edit/create configs...
>>>> 
>>>> K 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>      
>>>   
>>>  
>>>  
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>      
>   
>  
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anita.hall at simmortel.com  Mon Apr  9 18:43:35 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Mon, 9 Apr 2012 20:13:35 +0530
Subject: [Freeswitch-users] combining multiple SIP providers
Message-ID: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>

Hi

What is the best way to combine multiple SIP providers to achieve

1) Higher concurrency (for eg. 2 providers with 50 concurrent calling
limits could be combined to give a limit of 100)
2) Redundancy (use another if one is down)

I have a feeling that this will need some SIP Proxy like OpenSIPS but what
could be the architecture ?

Much thanks!

regards,
Anita
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From mitch.capper at gmail.com  Mon Apr  9 18:51:02 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 9 Apr 2012 07:51:02 -0700
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
References: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
Message-ID: <CALn233pM86xeb6me=3rwzsmZAn5C2hVk98irM7HRUBsA5sZTqw@mail.gmail.com>

I assume you mean for outgoing.  Redundancy is easy just place one
after another in a dialplan as if the first completes the call it will
break out of the dialplan and not hit the second, if the first fails
with continue on fail it will try the second.
As for higher concurrency if you don't need load based distribution
and round robin will work you could just alternate between the two
outgoing providers.  If you want to keep track of how many calls per
provider are active you could do so using scripting more easily (to
query active calls before routing) or try to keep track using a
database or the hash system (although I am not sure if you could write
atomic code for the hash system to keep the counts exactly active).

But it is definitely doable within FS without much hassle.

~mitch


From lazyvirus at gmx.com  Mon Apr  9 18:54:07 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Mon, 9 Apr 2012 16:54:07 +0200
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
References: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
Message-ID: <20120409165407.6df14646@osiris.defcon1>

On Mon, 9 Apr 2012 20:13:35 +0530
Anita Hall <anita.hall at simmortel.com> wrote:

> 
> I have a feeling that this will need some SIP Proxy like OpenSIPS but what
> could be the architecture ?

May be something like:
http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS
-- 
Hear about the young Chinese woman who just won the lottery?
One fortunate cookie...


From avi at avimarcus.net  Mon Apr  9 18:57:43 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 9 Apr 2012 17:57:43 +0300
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <20120409165407.6df14646@osiris.defcon1>
References: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
	<20120409165407.6df14646@osiris.defcon1>
Message-ID: <CALNZuwm8t99BV5wR4=1UL9qj3PnVB0uP9ikBM9VTaMGuV15pTg@mail.gmail.com>

As Mitch stated, opensips isn't necessary.
-Avi
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From bdfoster at endigotech.com  Mon Apr  9 19:25:02 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 9 Apr 2012 11:25:02 -0400
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CALNZuwm8t99BV5wR4=1UL9qj3PnVB0uP9ikBM9VTaMGuV15pTg@mail.gmail.com>
References: <CAGKPtBtyMoxohqas6LFt8v=yc1216mepjnpZdi-3f4ChtyNEEw@mail.gmail.com>
	<20120409165407.6df14646@osiris.defcon1>
	<CALNZuwm8t99BV5wR4=1UL9qj3PnVB0uP9ikBM9VTaMGuV15pTg@mail.gmail.com>
Message-ID: <CAKV-jFwdC5qCAfRce-uPJ5gDMOCtehEW6xiQwt4SFWgm40t4Uw@mail.gmail.com>

Is mod_distributor still being developed? That'll take care of you randomly
routing between the two (or more) gateways.

-BDF

On Mon, Apr 9, 2012 at 10:57 AM, Avi Marcus <avi at avimarcus.net> wrote:

> As Mitch stated, opensips isn't necessary.
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From krice at freeswitch.org  Mon Apr  9 19:28:01 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 09 Apr 2012 10:28:01 -0500
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CAKV-jFwdC5qCAfRce-uPJ5gDMOCtehEW6xiQwt4SFWgm40t4Uw@mail.gmail.com>
Message-ID: <CBA86A31.30E89%krice@freeswitch.org>

Theres that or mod_lcr or a number of ways to do it...

But if you are using > 1 carrier if they have varying rates per prefix you
probably want to go ahead and look at LCR so you can least cost route...
Depending on volume this may save you some money

K


On 4/9/12 10:25 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Is mod_distributor still being developed? That'll take care of you randomly
> routing between the two (or more) gateways.
> 
> -BDF
> 
> On Mon, Apr 9, 2012 at 10:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> As Mitch stated, opensips isn't necessary.
>> -Avi
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Mon Apr  9 19:36:23 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 9 Apr 2012 11:36:23 -0400
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CBA86A31.30E89%krice@freeswitch.org>
References: <CAKV-jFwdC5qCAfRce-uPJ5gDMOCtehEW6xiQwt4SFWgm40t4Uw@mail.gmail.com>
	<CBA86A31.30E89%krice@freeswitch.org>
Message-ID: <CAKV-jFw_3AcwYCARuKVVNCznZfovr2PEVnU9MCSVwvpee5H3kQ@mail.gmail.com>

Agreed. mod_lcr used here w/ 15ish carriers. Seems like no two carriers
ever have the same rate on the same route.

-BDF

On Mon, Apr 9, 2012 at 11:28 AM, Ken Rice <krice at freeswitch.org> wrote:

>  Theres that or mod_lcr or a number of ways to do it...
>
> But if you are using > 1 carrier if they have varying rates per prefix you
> probably want to go ahead and look at LCR so you can least cost route...
> Depending on volume this may save you some money
>
> K
>
>
>
> On 4/9/12 10:25 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
> Is mod_distributor still being developed? That'll take care of you
> randomly routing between the two (or more) gateways.
>
> -BDF
>
> On Mon, Apr 9, 2012 at 10:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
> As Mitch stated, opensips isn't necessary.
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From davidjbrazier at gmail.com  Mon Apr  9 19:47:13 2012
From: davidjbrazier at gmail.com (David Brazier)
Date: Mon, 9 Apr 2012 16:47:13 +0100
Subject: [Freeswitch-users] mod_avmd false positives, ineffecient
Message-ID: <CAGpgTK_dH424YC4VaUrknyyLTfWhnKdypP2qsBNvQAmC7LjmUA@mail.gmail.com>

Hi

I've been testing mod_avmd and got a lot of false positives - i.e. normal
speech detected as beep.  As I understand it, the basic approach is to look
for audio with one very dominant frequency - the beep.  It does this by
getting a frequency estimate and then calculats the amplitude of the audio
at that frequency, looking for a consistent run of high amplitudes.  But
I've found a few things that are either not right or I have misunderstood:

1.  The expensive desa2 frequency estimate is calculated on every sample,
and worse, as each frame arrives (say 160 samples) it calculates the desa2
for every sample in the circular buffer - 800 samples (rounded up to 1024 I
think?).  So that is using excessive CPU, and anyway I don't think the
frequency estimate should be calculated on every sample, because in normal
speech, the frequency is changing and so successive samples might register
a high amplitude but with different frequencies, causing false positives.
2.  The amplitude function doesn't appear to produce a constant result when
the input audio is a pure tone, although the deas2 frequency estimate is
close.
3.  The running mean sma_b is tested against zero - which basically means
any positive amplitude meeting the "valid" cases will count as a detected
beep.

I have re-worked mod_avmd.c to:

1.  Calculate the frequency only once every 10ms (sine len).
2.  Calculate the running variance of frequency over the required beep
length (100ms) - ie. last 10 estimates.

This is producing good results - would welcome comment & if anyone else can
try it that would be great.  Could be further tided up as it doesn't really
require a circular buffer for samples at all - could just do the desa2
calculation direct from the sample buffer, at least once per frame (2 per
frame for 8000kHz, 20ms frames).  Also, goertzel.c isn't used by the module
at all, and amplitude.c isn't used in my version and can be removed.

Raised as http://jira.freeswitch.org/browse/FS-4090 with proposed patch.

Regards

David
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From manjiri05_deshpande at yahoo.co.in  Mon Apr  9 19:52:28 2012
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Mon, 9 Apr 2012 23:52:28 +0800 (SGT)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1333986748.98579.BPMail_high_noncarrier@web192502.mail.sg3.yahoo.com>


http://elastomer-kazan.ru/entrance.php?wuluhaxynj=56&jvabyxuv=43


From anita.hall at simmortel.com  Mon Apr  9 20:03:24 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Mon, 9 Apr 2012 21:33:24 +0530
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CAKV-jFw_3AcwYCARuKVVNCznZfovr2PEVnU9MCSVwvpee5H3kQ@mail.gmail.com>
References: <CAKV-jFwdC5qCAfRce-uPJ5gDMOCtehEW6xiQwt4SFWgm40t4Uw@mail.gmail.com>
	<CBA86A31.30E89%krice@freeswitch.org>
	<CAKV-jFw_3AcwYCARuKVVNCznZfovr2PEVnU9MCSVwvpee5H3kQ@mail.gmail.com>
Message-ID: <CAGKPtBsmWA9i1LmoF16Fa1AXDXDQZV7h3c-6a7y15T5Nxc+jQw@mail.gmail.com>

Thanks everybody! Much appreciated :)

mod_lcr is what I am trying first!

regards,
Anita



On Mon, Apr 9, 2012 at 9:06 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Agreed. mod_lcr used here w/ 15ish carriers. Seems like no two carriers
> ever have the same rate on the same route.
>
> -BDF
>
>
> On Mon, Apr 9, 2012 at 11:28 AM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Theres that or mod_lcr or a number of ways to do it...
>>
>> But if you are using > 1 carrier if they have varying rates per prefix
>> you probably want to go ahead and look at LCR so you can least cost
>> route... Depending on volume this may save you some money
>>
>> K
>>
>>
>>
>> On 4/9/12 10:25 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>
>> Is mod_distributor still being developed? That'll take care of you
>> randomly routing between the two (or more) gateways.
>>
>> -BDF
>>
>> On Mon, Apr 9, 2012 at 10:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> As Mitch stated, opensips isn't necessary.
>> -Avi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From dujinfang at gmail.com  Mon Apr  9 20:42:57 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 10 Apr 2012 00:42:57 +0800
Subject: [Freeswitch-users] Open Bugs on Jira and Call for help
In-Reply-To: <CBA847DA.30E76%krice@freeswitch.org>
References: <CBA847DA.30E76%krice@freeswitch.org>
Message-ID: <85BBCF7888C542FF98416D4C2C94248E@gmail.com>

Thanks. I updated the wiki a bit.

I haven't try the video-bridge and video-floor-only, will try some other day when I have plenty of video phones. Or anyone knows what it's mean and update wiki?  

On Monday, April 9, 2012 at 9:01 PM, Ken Rice wrote:

> Re: [Freeswitch-users] Open Bugs on Jira and Call for help The stuff in mod_XXX/conf should be exhaustive for the module...
>  
> The stuff in conf/${style}/* should be a working config specific to the example being presented in ${style}
>  
>  
> As far as wiki differening from code, I think we should go with what the code says...   There are a few cases where this might not be true (in the even of say a regression)
>  
> K
>  
>  
> On 4/9/12 7:48 AM, "Seven Du" <dujinfang at gmail.com> wrote:
>  
> >  
> > Thanks Ken.
> >  
> > see http://jira.freeswitch.org/browse/FS-4088
> >  
> > questions:
> >  
> > 1) I see confs both in mod_xxx/conf and conf/vanilla, should I modify both? I might miss the vanilla story, would anyone tell it briefly?
> >  
> > 2) wiki saids http://wiki.freeswitch.org/wiki/Mod_conference
> >  
> > conference-flags Can be any combination of wait-mod and waste-bandwidth. wait-mod  
> >  
> > but I only see  
> >  
> > for (i = 0; i < argc && argv[i]; i++) {
> > if (!strcasecmp(argv[i], "wait-mod")) {
> > *f |= CFLAG_WAIT_MOD;
> > } else if (!strcasecmp(argv[i], "video-floor-only")) {
> > *f |= CFLAG_VID_FLOOR;
> > } else if (!strcasecmp(argv[i], "video-bridge")) {
> > *f |= CFLAG_VIDEO_BRIDGE;
> > }
> >  
> > is wiki wrong or should we patch mod_conference.c to enable CFLAG_WASTE_FLAG ?  I see this flag is not actually used.
> >  
> > 7.
> >    
> >  
> > On Thursday, March 29, 2012 at 12:59 AM, Ken Rice wrote:
> >  
> > >  
> > > Re: [Freeswitch-users] Open Bugs on Jira and Call for help Actually we can opena  tracking bug on Jira or just email me the diffs I can review and commit that way...
> > >  
> > > Chjeck out git-format-patch its a good way to format a patch so you get credit also
> > >  
> > > K
> > >  
> > >  
> > > On 3/28/12 11:41 AM, "Seven Du" <dujinfang at gmail.com> wrote:
> > >  
> > > >  
> > > > Cool. I'll update some of my open bugs on jira. And for documentation help, I suggest also address the indents, especially don't mix spaces and tabs, I guess all like tabs? Also, is there any coordination of which conf files who's working on or should all choose randomly? And show we commit to git or report on jira?
> > > >  
> > > >    
> > > >  
> > > > On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote:
> > > >  
> > > > >  
> > > > > Open Bugs on Jira and Call for help If you have open bugs on Jira, please make sure they are up to date on any all information....  
> > > > >  
> > > > > If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We?re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible...
> > > > >  
> > > > > Also, I?m looking for some configuration documentation help. If you?ve pulled git in the past few days you?ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are.  The end goal is to get all the options documented in their default states.  
> > > > > What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs...  
> > > > >  
> > > > > K  
> > > > > _________________________________________________________________________
> > > > > Professional FreeSWITCH Consulting Services:
> > > > > consulting at freeswitch.org
> > > > > http://www.freeswitchsolutions.com
> > > > >  
> > > > > 
> > > > > 
> > > > >  
> > > > > Official FreeSWITCH Sites
> > > > > http://www.freeswitch.org
> > > > > http://wiki.freeswitch.org
> > > > > http://www.cluecon.com
> > > > >  
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > > >       
> > > >   
> > > >   
> > > >   
> > > >  
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org
> > > > http://www.freeswitchsolutions.com
> > > >  
> > > > 
> > > > 
> > > >  
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > >  
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >  
> > > 
> > > 
> > >  
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >  
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >       
> >   
> >   
> >   
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >  
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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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From tech at fahrwerk-berlin.de  Mon Apr  9 20:22:08 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Mon, 09 Apr 2012 18:22:08 +0200
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones when
 phone is ringing; possible?
Message-ID: <4F830CB0.6020301@fahrwerk-berlin.de>

Hi all,

Is it possible to use moh and if a phone is ringing to play ring tones 
to the calling party?

Thanks in advance,
Georg


From tech at fahrwerk-berlin.de  Mon Apr  9 20:25:28 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Mon, 09 Apr 2012 18:25:28 +0200
Subject: [Freeswitch-users] alternative to mod_hash which is persistent
Message-ID: <4F830D78.4010001@fahrwerk-berlin.de>

Hi all,

As the wiki writes, that mod_hash "provides a non-persistent datastore 
for limit data. This data is specific to a particular FreeSWITCH 
instance", I'm searching for an alternative.

I would like to setup a callforwarding extension, and using mod_hash at 
the moment for storing the target number. After a reboot, of course, the 
hash is gone.

Could someone point me at an alternative?

Thanks,
Georg


From msc at freeswitch.org  Mon Apr  9 21:18:02 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Apr 2012 10:18:02 -0700
Subject: [Freeswitch-users] mod_avmd false positives, ineffecient
In-Reply-To: <CAGpgTK_dH424YC4VaUrknyyLTfWhnKdypP2qsBNvQAmC7LjmUA@mail.gmail.com>
References: <CAGpgTK_dH424YC4VaUrknyyLTfWhnKdypP2qsBNvQAmC7LjmUA@mail.gmail.com>
Message-ID: <CAKzWOxXfSm7pZzs5CVrq2N4W+=PPvNd03a8do9Hkd0Agv_mXeQ@mail.gmail.com>

Thank you for taking the time not only to make code improvements but also
to inform the community of what you did and why. I suspect that EricDC is
going to be the one who is in the best position to comment on the changes
since he's the author of mod_avmd. In the meantime, though, if any out
there is using mod_avmd and is in a position to try out David's changes and
report back on jira then that would be most appreciated.

Thanks,
MC

On Mon, Apr 9, 2012 at 8:47 AM, David Brazier <davidjbrazier at gmail.com>wrote:

> Hi
>
> I've been testing mod_avmd and got a lot of false positives - i.e. normal
> speech detected as beep.  As I understand it, the basic approach is to look
> for audio with one very dominant frequency - the beep.  It does this by
> getting a frequency estimate and then calculats the amplitude of the audio
> at that frequency, looking for a consistent run of high amplitudes.  But
> I've found a few things that are either not right or I have misunderstood:
>
> 1.  The expensive desa2 frequency estimate is calculated on every sample,
> and worse, as each frame arrives (say 160 samples) it calculates the desa2
> for every sample in the circular buffer - 800 samples (rounded up to 1024 I
> think?).  So that is using excessive CPU, and anyway I don't think the
> frequency estimate should be calculated on every sample, because in normal
> speech, the frequency is changing and so successive samples might register
> a high amplitude but with different frequencies, causing false positives.
> 2.  The amplitude function doesn't appear to produce a constant result
> when the input audio is a pure tone, although the deas2 frequency estimate
> is close.
> 3.  The running mean sma_b is tested against zero - which basically means
> any positive amplitude meeting the "valid" cases will count as a detected
> beep.
>
> I have re-worked mod_avmd.c to:
>
> 1.  Calculate the frequency only once every 10ms (sine len).
> 2.  Calculate the running variance of frequency over the required beep
> length (100ms) - ie. last 10 estimates.
>
> This is producing good results - would welcome comment & if anyone else
> can try it that would be great.  Could be further tided up as it doesn't
> really require a circular buffer for samples at all - could just do the
> desa2 calculation direct from the sample buffer, at least once per frame (2
> per frame for 8000kHz, 20ms frames).  Also, goertzel.c isn't used by the
> module at all, and amplitude.c isn't used in my version and can be removed.
>
> Raised as http://jira.freeswitch.org/browse/FS-4090 with proposed patch.
>
> Regards
>
> David
>
>
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From msc at freeswitch.org  Mon Apr  9 21:20:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Apr 2012 10:20:18 -0700
Subject: [Freeswitch-users] alternative to mod_hash which is persistent
In-Reply-To: <4F830D78.4010001@fahrwerk-berlin.de>
References: <4F830D78.4010001@fahrwerk-berlin.de>
Message-ID: <CAKzWOxUWg4VdtzJpPq5EZn7j36zBK3W6SH3_nEC-tU5pwf2hZQ@mail.gmail.com>

Perhaps this would do it?
http://wiki.freeswitch.org/wiki/Mod_db

-MC

On Mon, Apr 9, 2012 at 9:25 AM, Fahrwerk Kurierkollektiv (Technik) <
tech at fahrwerk-berlin.de> wrote:

> Hi all,
>
> As the wiki writes, that mod_hash "provides a non-persistent datastore
> for limit data. This data is specific to a particular FreeSWITCH
> instance", I'm searching for an alternative.
>
> I would like to setup a callforwarding extension, and using mod_hash at
> the moment for storing the target number. After a reboot, of course, the
> hash is gone.
>
> Could someone point me at an alternative?
>
> Thanks,
> Georg
>
>
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From djbinter at gmail.com  Mon Apr  9 21:34:52 2012
From: djbinter at gmail.com (DJB International)
Date: Mon, 9 Apr 2012 10:34:52 -0700
Subject: [Freeswitch-users] BLF Presence
In-Reply-To: <-159227813343798664@unknownmsgid>
References: <CAAB3jWujuvEp=6k_=smUCO7OVp5nTDD8DdG5GYrrQ7rJbUgAZw@mail.gmail.com>
	<CAAtWKeEOgRtzyO+icNtizt2VtWZKi44k6nrNQx=KcK+itTz=vA@mail.gmail.com>
	<-159227813343798664@unknownmsgid>
Message-ID: <CAAB3jWsuidDBRLDGemeAdfhBcX-AdPBJFgciYS2V_=Jcog38TQ@mail.gmail.com>

Any other advices regarding sip_presence table?

Thank you.


On Sat, Apr 7, 2012 at 1:22 AM, Dorn DJBinter <djbinter at gmail.com> wrote:

> Thank you, Yehavi.
>
> Presence is working fine.  I just want to see the status in sip_presence
> table which I assume that it would have similar info as "sip show
> subscriptions", but on my FreeSWITCH system, that table does not populate
> any info at all which I wonder whether I need to adjust any variables
> somewhere.
>
>
>
> On Apr 6, 2012, at 11:41 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
> Hi,
>
>   Are you asking that because presence does not work, or just because you
> want to see the presence tables?
>
> Note that turnning on the presence feature is not enough. You have to
> define a speed dial ("buddy" in Polycom's jargon) to subscribe for
> presence. You should have BW (buddy watch) enabled for each speed dial for
> it to subscribe for presence.
>
>                            Regards, __Yehavi:
>
> 2012/4/7 DJB International <djbinter at gmail.com>
>
>> I currently have my Polycom 650 phone set up with presence enabled:
>>
>> <feature feature.1.name="presence" feature.1.enabled="1" />
>>
>>
>>
>> However, when I looked at the table sip_presence in
>> sofia_reg_internal.db, I do not see any information in there.
>>
>> What I am trying to achieve is something similar to "sip show
>> subscription" to verify BLF subscriptions.
>>
>> Is there something that I am missing here.  Please advise.
>>
>> Thank you.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Mon Apr  9 21:39:29 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 9 Apr 2012 10:39:29 -0700
Subject: [Freeswitch-users] Developers is this a bridge change or Jira/Bug
	issue?
Message-ID: <F660A0B2-57A4-4077-86D7-EE9E54D7B750@mgtech.com>

Testing the recent gits introduced a problem which I was able to work around but wondered if this is an intended change or bug: If the caller_id_number is one of the numbers in a bridge command only 1 ring is done and the bridge ends, it appears as if originate_timeout is ignored, the trace is normal. Prior to the git update the bridge would continue to ring to the timeout and the caller would be bridged to all the numbers in the bridge so any phone could answer the call.

What used to work in older git:
            <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/>

Newer git. Now I have to execute an extension to check who is calling so they are not in the bridge:
    <extension name="bridge_call">
      <condition field="destination_number" expression="^bridge_call$">
      </condition>   
        <condition field="${caller_id_number_work}" expression="^9161112222$" break="on-true">
            <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444"/>
        </condition>
        <condition field="${caller_id_number_work}" expression="^9163334444$" break="on-true">
            <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222"/>
        </condition>
        <condition field="${caller_id_number_work}" expression="^(\d+)$">
            <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/>
        </condition>
    </extension>

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From nestor at tiendalinux.com  Mon Apr  9 23:03:57 2012
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Mon, 09 Apr 2012 14:03:57 -0500
Subject: [Freeswitch-users] dinamically change sip_from_user in the dialplan
Message-ID: <4F83329D.9000702@tiendalinux.com>

Hi Guys,

I want to know if is possible to change the sip_from_user at the dialplan.

My current pstn sip provider needs the caller id to be set to one of the 
numbers assigned to me, this way i can place the call using the 
following options at the dialplan:

<extension name="local">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^9(\d{7})$">
<action application="set" data="effective_caller_id_number=12345678"/>
<action application="bridge" data="sofia/profile/$1 at sipprovider.com"/>
</condition>
</extension>

The above works, also if i define a gateway as the following:

<include>
<gateway name="myprovider">
<param name="from-user" value="12345678"/>
<param name="proxy" value="sipprovider.com"/>
<param name="register" value="false"/>
<param name="ping" value="25"/>
</gateway>
</include>

and place the call using:

<action application="bridge" data="sofia/gateway/myprovider/$1"/>

the above woks too, but i am tied to the from_user i have defined.

So i wanted to use be able to change the from_user dinamically according 
to my needs at the dialplan level, is that possible ? i have tried the 
following but it doesn't work

<action application="bridge" 
data="{sip_from_user=12345678}sofia/gateway/myprovider/$1"/>

Any suggestions ? or is not possible to define a gateway and change some 
of its parameters in the dialplan ? i just want to use the ping 
functionality and have something more organized.

Slds.

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia



From vipkilla at gmail.com  Mon Apr  9 23:10:22 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 9 Apr 2012 15:10:22 -0400
Subject: [Freeswitch-users] SCA/SLA not working with Polycoms on inbound
	calls
Message-ID: <CAC-LwPMmqNzqJ-GFQ=42mAoQtfOHZ5Y3Gwod3ATXFxRFZOaJRQ@mail.gmail.com>

I have SCA/SLA working with Polycom phones (firmware 3.3.1) on
outbound calls. But inbound is not working. I have read over
http://wiki.freeswitch.org/wiki/Shared_Line_Appearance and followed
the instructions but still no luck. I have the following in my sofia
profile:
<param name="manage-shared-appearance" value="true"/>
<param name="multiple-registrations" value="true"/>
<param name="manage-presence" value="true"/>

I have 2 Polycom phones registered (1000 & 1004) and the second line
on each phone is a shared line, 9000.
A call comes in to 9000 and both phones ring as expected but once one
of them picks up, the line 2 on the other phone has no affect on the
call (normally it would allow 'barge-in' or it would take the call if
it was on hold). On outbound, everything works so I do not believe it
is a Polycom configuration issue.

Here is the dialplan bridge I'm using:

<action application="bridge" data="${sofia_contact(9000 at thedomain.com)}"/>

Any suggestions would be greatly appreciated, thanks.


From msc at freeswitch.org  Mon Apr  9 23:20:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 9 Apr 2012 12:20:55 -0700
Subject: [Freeswitch-users] Developers is this a bridge change or
	Jira/Bug issue?
In-Reply-To: <F660A0B2-57A4-4077-86D7-EE9E54D7B750@mgtech.com>
References: <F660A0B2-57A4-4077-86D7-EE9E54D7B750@mgtech.com>
Message-ID: <CAKzWOxWr7dNdCOzqoOQZPR2Yakq12-+oQf8veXqCTALXgcKq0Q@mail.gmail.com>

I would like to see the console debug log and corresponding sip trace on a
working call vs. non-working call. Does it only fail if the destination
number is the same as the origination_caller_id_number, or are there other
scenarios where it works (or doesn't work)? Throw that up on pb and let's
have a look.

-MC

On Mon, Apr 9, 2012 at 10:39 AM, Mario G <mario_fs at mgtech.com> wrote:

> Testing the recent gits introduced a problem which I was able to work
> around but wondered if this is an intended change or bug: If the
> caller_id_number is one of the numbers in a bridge command only 1 ring is
> done and the bridge ends, it appears as if originate_timeout is ignored,
> the trace is normal. Prior to the git update the bridge would continue to
> ring to the timeout and the caller would be bridged to all the numbers in
> the bridge so any phone could answer the call.
>
> What used to work in older git:
>             <action application="bridge" data=
> "<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@
> ${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/
> 19163334444"/>
>
> Newer git. Now I have to execute an extension to check who is calling so
> they are not in the bridge:
>     <extension name="bridge_call">
>       <condition field="destination_number" expression="^bridge_call$">
>       </condition>
>         <condition field="${caller_id_number_work}" expression=
> "^9161112222$" break="on-true">
>             <action application="bridge" data=
> "<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@
> ${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444"/>
>         </condition>
>         <condition field="${caller_id_number_work}" expression=
> "^9163334444$" break="on-true">
>             <action application="bridge" data=
> "<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@
> ${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222"/>
>         </condition>
>         <condition field="${caller_id_number_work}" expression="^(\d+)$">
>             <action application="bridge" data=
> "<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@
> ${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/
> 19163334444"/>
>         </condition>
>     </extension>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vipkilla at gmail.com  Mon Apr  9 23:42:07 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 9 Apr 2012 15:42:07 -0400
Subject: [Freeswitch-users] SCA/SLA not working with Polycoms on inbound
	calls
In-Reply-To: <CAC-LwPMmqNzqJ-GFQ=42mAoQtfOHZ5Y3Gwod3ATXFxRFZOaJRQ@mail.gmail.com>
References: <CAC-LwPMmqNzqJ-GFQ=42mAoQtfOHZ5Y3Gwod3ATXFxRFZOaJRQ@mail.gmail.com>
Message-ID: <CAC-LwPO0ac-URuLCKn9TtkBCHq1=j91F=DM62T5ZuWUpJh7_oA@mail.gmail.com>

FYI, i got this working, apparently {sip_invite_domain=thedomain.com}
is needed such as:
<action application="bridge"
data="{sip_invite_domain=thedomain.com}${sofia_contact(9000 at thedomain.com)}"/>

Thanks anyway!

On Mon, Apr 9, 2012 at 3:10 PM, Vik Killa <vipkilla at gmail.com> wrote:
> I have SCA/SLA working with Polycom phones (firmware 3.3.1) on
> outbound calls. But inbound is not working. I have read over
> http://wiki.freeswitch.org/wiki/Shared_Line_Appearance and followed
> the instructions but still no luck. I have the following in my sofia
> profile:
> <param name="manage-shared-appearance" value="true"/>
> <param name="multiple-registrations" value="true"/>
> <param name="manage-presence" value="true"/>
>
> I have 2 Polycom phones registered (1000 & 1004) and the second line
> on each phone is a shared line, 9000.
> A call comes in to 9000 and both phones ring as expected but once one
> of them picks up, the line 2 on the other phone has no affect on the
> call (normally it would allow 'barge-in' or it would take the call if
> it was on hold). On outbound, everything works so I do not believe it
> is a Polycom configuration issue.
>
> Here is the dialplan bridge I'm using:
>
> <action application="bridge" data="${sofia_contact(9000 at thedomain.com)}"/>
>
> Any suggestions would be greatly appreciated, thanks.


From anthony.minessale at gmail.com  Mon Apr  9 23:49:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 9 Apr 2012 14:49:15 -0500
Subject: [Freeswitch-users] SCA/SLA not working with Polycoms on inbound
	calls
In-Reply-To: <CAC-LwPO0ac-URuLCKn9TtkBCHq1=j91F=DM62T5ZuWUpJh7_oA@mail.gmail.com>
References: <CAC-LwPMmqNzqJ-GFQ=42mAoQtfOHZ5Y3Gwod3ATXFxRFZOaJRQ@mail.gmail.com>
	<CAC-LwPO0ac-URuLCKn9TtkBCHq1=j91F=DM62T5ZuWUpJh7_oA@mail.gmail.com>
Message-ID: <CAKbxfG_aG5iPSOfwvXdRZk60Qkn8NG0zd-xRm1u==aK8Jw17eQ@mail.gmail.com>

also same effect if you use IP instead of domains in your phone
configs which is recommended.

On Mon, Apr 9, 2012 at 2:42 PM, Vik Killa <vipkilla at gmail.com> wrote:
> FYI, i got this working, apparently {sip_invite_domain=thedomain.com}
> is needed such as:
> <action application="bridge"
> data="{sip_invite_domain=thedomain.com}${sofia_contact(9000 at thedomain.com)}"/>
>
> Thanks anyway!
>
> On Mon, Apr 9, 2012 at 3:10 PM, Vik Killa <vipkilla at gmail.com> wrote:
>> I have SCA/SLA working with Polycom phones (firmware 3.3.1) on
>> outbound calls. But inbound is not working. I have read over
>> http://wiki.freeswitch.org/wiki/Shared_Line_Appearance and followed
>> the instructions but still no luck. I have the following in my sofia
>> profile:
>> <param name="manage-shared-appearance" value="true"/>
>> <param name="multiple-registrations" value="true"/>
>> <param name="manage-presence" value="true"/>
>>
>> I have 2 Polycom phones registered (1000 & 1004) and the second line
>> on each phone is a shared line, 9000.
>> A call comes in to 9000 and both phones ring as expected but once one
>> of them picks up, the line 2 on the other phone has no affect on the
>> call (normally it would allow 'barge-in' or it would take the call if
>> it was on hold). On outbound, everything works so I do not believe it
>> is a Polycom configuration issue.
>>
>> Here is the dialplan bridge I'm using:
>>
>> <action application="bridge" data="${sofia_contact(9000 at thedomain.com)}"/>
>>
>> Any suggestions would be greatly appreciated, thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
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From vipkilla at gmail.com  Mon Apr  9 23:55:09 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 9 Apr 2012 15:55:09 -0400
Subject: [Freeswitch-users] SCA/SLA not working with Polycoms on inbound
	calls
In-Reply-To: <CAKbxfG_aG5iPSOfwvXdRZk60Qkn8NG0zd-xRm1u==aK8Jw17eQ@mail.gmail.com>
References: <CAC-LwPMmqNzqJ-GFQ=42mAoQtfOHZ5Y3Gwod3ATXFxRFZOaJRQ@mail.gmail.com>
	<CAC-LwPO0ac-URuLCKn9TtkBCHq1=j91F=DM62T5ZuWUpJh7_oA@mail.gmail.com>
	<CAKbxfG_aG5iPSOfwvXdRZk60Qkn8NG0zd-xRm1u==aK8Jw17eQ@mail.gmail.com>
Message-ID: <CAC-LwPNFCh33u1NxX-QOTsYGXD-H7oGL6Tg+ZdGBZSXSLPQykg@mail.gmail.com>

On Mon, Apr 9, 2012 at 3:49 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> also same effect if you use IP instead of domains in your phone
> configs which is recommended.

Well that's not really possible in multi-tenant, is it?


From bdfoster at endigotech.com  Tue Apr 10 00:55:35 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 9 Apr 2012 16:55:35 -0400
Subject: [Freeswitch-users] dinamically change sip_from_user in the
	dialplan
In-Reply-To: <4F83329D.9000702@tiendalinux.com>
References: <4F83329D.9000702@tiendalinux.com>
Message-ID: <CAKV-jFxBb4GowH+1f3Qy=f=+-kZ-GwH=WmS+DakRzTYS=jjA+Q@mail.gmail.com>

When you set the variable or param in the gateway configuration it
overrides variables set elsewhere in the call. Try removing the variable in
the gateway configuration and setting that in the dialplan before bridging
the call.

-BDF
On Apr 9, 2012 3:05 PM, "Nestor A Diaz" <nestor at tiendalinux.com> wrote:

> Hi Guys,
>
> I want to know if is possible to change the sip_from_user at the dialplan.
>
> My current pstn sip provider needs the caller id to be set to one of the
> numbers assigned to me, this way i can place the call using the
> following options at the dialplan:
>
> <extension name="local">
> <condition field="${toll_allow}" expression="local"/>
> <condition field="destination_number" expression="^9(\d{7})$">
> <action application="set" data="effective_caller_id_number=12345678"/>
> <action application="bridge" data="sofia/profile/$1 at sipprovider.com"/>
> </condition>
> </extension>
>
> The above works, also if i define a gateway as the following:
>
> <include>
> <gateway name="myprovider">
> <param name="from-user" value="12345678"/>
> <param name="proxy" value="sipprovider.com"/>
> <param name="register" value="false"/>
> <param name="ping" value="25"/>
> </gateway>
> </include>
>
> and place the call using:
>
> <action application="bridge" data="sofia/gateway/myprovider/$1"/>
>
> the above woks too, but i am tied to the from_user i have defined.
>
> So i wanted to use be able to change the from_user dinamically according
> to my needs at the dialplan level, is that possible ? i have tried the
> following but it doesn't work
>
> <action application="bridge"
> data="{sip_from_user=12345678}sofia/gateway/myprovider/$1"/>
>
> Any suggestions ? or is not possible to define a gateway and change some
> of its parameters in the dialplan ? i just want to use the ping
> functionality and have something more organized.
>
> Slds.
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Tue Apr 10 00:57:26 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 9 Apr 2012 16:57:26 -0400
Subject: [Freeswitch-users] dinamically change sip_from_user in the
	dialplan
In-Reply-To: <CAKV-jFxBb4GowH+1f3Qy=f=+-kZ-GwH=WmS+DakRzTYS=jjA+Q@mail.gmail.com>
References: <4F83329D.9000702@tiendalinux.com>
	<CAKV-jFxBb4GowH+1f3Qy=f=+-kZ-GwH=WmS+DakRzTYS=jjA+Q@mail.gmail.com>
Message-ID: <CAKV-jFz+BfJ4Sxf2U4+tQGXFTehwgCXg2pb6jrL5YJZPwho_cw@mail.gmail.com>

Also use the dialplan application "set" to set that before you do the
actual bridge not with the bridge (personal preference).
On Apr 9, 2012 4:55 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> When you set the variable or param in the gateway configuration it
> overrides variables set elsewhere in the call. Try removing the variable in
> the gateway configuration and setting that in the dialplan before bridging
> the call.
>
> -BDF
> On Apr 9, 2012 3:05 PM, "Nestor A Diaz" <nestor at tiendalinux.com> wrote:
>
>> Hi Guys,
>>
>> I want to know if is possible to change the sip_from_user at the dialplan.
>>
>> My current pstn sip provider needs the caller id to be set to one of the
>> numbers assigned to me, this way i can place the call using the
>> following options at the dialplan:
>>
>> <extension name="local">
>> <condition field="${toll_allow}" expression="local"/>
>> <condition field="destination_number" expression="^9(\d{7})$">
>> <action application="set" data="effective_caller_id_number=12345678"/>
>> <action application="bridge" data="sofia/profile/$1 at sipprovider.com"/>
>> </condition>
>> </extension>
>>
>> The above works, also if i define a gateway as the following:
>>
>> <include>
>> <gateway name="myprovider">
>> <param name="from-user" value="12345678"/>
>> <param name="proxy" value="sipprovider.com"/>
>> <param name="register" value="false"/>
>> <param name="ping" value="25"/>
>> </gateway>
>> </include>
>>
>> and place the call using:
>>
>> <action application="bridge" data="sofia/gateway/myprovider/$1"/>
>>
>> the above woks too, but i am tied to the from_user i have defined.
>>
>> So i wanted to use be able to change the from_user dinamically according
>> to my needs at the dialplan level, is that possible ? i have tried the
>> following but it doesn't work
>>
>> <action application="bridge"
>> data="{sip_from_user=12345678}sofia/gateway/myprovider/$1"/>
>>
>> Any suggestions ? or is not possible to define a gateway and change some
>> of its parameters in the dialplan ? i just want to use the ping
>> functionality and have something more organized.
>>
>> Slds.
>>
>> --
>> Nestor A. Diaz
>> Ingeniero de Sistemas
>> Tel. +57 1-485-3020 x 211
>> Cel. +57 316-227-3593
>> Tel. SIP: sip:211 at tiendalinux.com
>> Email/MSN: nestor at tiendalinux.com
>> http://www.tiendalinux.com/
>> Bogota, Colombia
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From jean.marc.hyppolite at gmail.com  Tue Apr 10 01:07:22 2012
From: jean.marc.hyppolite at gmail.com (Jean-Marc Hyppolite)
Date: Mon, 9 Apr 2012 17:07:22 -0400
Subject: [Freeswitch-users]  Help with tone detection
Message-ID: <4f834f8d.48bbe00a.39b4.19b7@mx.google.com>

Hello everyone,

 

Is it possible to have some help in setting up mod_spandsp for call progress
tone detection? I have not been successful thus far. I keep getting an
"error reading frame" message like this one below.

 

============================================================================
===================

2012-04-09 16:03:22.550303 [DEBUG] mod_spandsp_dsp.c:379
(sofia/internal/sip:1000 at 184.162.247.104:32986) Starting tone detection for
'1'

2012-04-09 16:03:22.550303 [INFO] mod_spandsp_dsp.c:411
(sofia/internal/sip:1000 at 184.162.247.104:32986) initializing tone detector

2012-04-09 16:03:22.550303 [DEBUG] switch_core_media_bug.c:462 Attaching BUG
to sofia/internal/sip:1000 at 184.162.247.104:32986

2012-04-09 16:03:22.590301 [DEBUG] switch_core_io.c:340 Setting BUG Codec
PCMU:0

2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:422
(sofia/internal/sip:1000 at 184.162.247.104:32986) error reading frame

2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:447
(sofia/internal/sip:1000 at 184.162.247.104:32986) destroying tone detector

============================================================================
===================

 

My spandsp.conf.xml file:

 

============================================================================
===================

<configuration name="spandsp.conf" description="Tone detector descriptors">

   <descriptors>

     <!-- These tones are defined in Annex to ITU Operational Bulletin No.
781 - 1.II.2003 -->

     <!-- Various Tones Used in National Networks (According to ITU-T
Recommendation E.180)(03/1998) -->

 

     <!-- North America -->

     <descriptor name="1">

       <tone name="CED_TONE">

         <element freq1="2100" freq2="0" min="500" max="0"/>

       </tone>

       <tone name="SIT">

         <element freq1="950" freq2="0" min="256" max="400"/>

         <element freq1="1400" freq2="0" min="256" max="400"/>

         <element freq1="1800" freq2="0" min="256" max="400"/>

       </tone>

       <tone name="REORDER_TONE">

         <element freq1="480" freq2="620" min="224" max="272"/>

         <element freq1="0" freq2="0" min="224" max="272"/>

       </tone>

       <tone name="BUSY_TONE">

         <element freq1="480" freq2="620" min="464" max="516"/>

         <element freq1="0" freq2="0" min="464" max="516"/>

       </tone>

     </descriptor>

   </descriptors>

</configuration>

============================================================================
===================

 

Thank you for your help.

 

Jean-Marc.

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From nestor at tiendalinux.com  Tue Apr 10 01:21:18 2012
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Mon, 09 Apr 2012 16:21:18 -0500
Subject: [Freeswitch-users] [SOLVED] Re: dinamically change sip_from_user in
 the dialplan
In-Reply-To: <4F83329D.9000702@tiendalinux.com>
References: <4F83329D.9000702@tiendalinux.com>
Message-ID: <4F8352CE.2090502@tiendalinux.com>

Hi, the problem was solved, the question have been already answered by 
someone on this list, the solution is to put:

+ <param name="caller-id-in-from" value="true"/>

in the gateway profile.

Slds.

-- 

Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia


On 04/09/2012 02:03 PM, Nestor A Diaz wrote:
> Hi Guys,
>
> I want to know if is possible to change the sip_from_user at the dialplan.
>
> My current pstn sip provider needs the caller id to be set to one of the
> numbers assigned to me, this way i can place the call using the
> following options at the dialplan:
>
> <extension name="local">
> <condition field="${toll_allow}" expression="local"/>
> <condition field="destination_number" expression="^9(\d{7})$">
> <action application="set" data="effective_caller_id_number=12345678"/>
> <action application="bridge" data="sofia/profile/$1 at sipprovider.com"/>
> </condition>
> </extension>
>
> The above works, also if i define a gateway as the following:
>
> <include>
> <gateway name="myprovider">
> <param name="from-user" value="12345678"/>
> <param name="proxy" value="sipprovider.com"/>
> <param name="register" value="false"/>
> <param name="ping" value="25"/>
> </gateway>
> </include>
>
> and place the call using:
>
> <action application="bridge" data="sofia/gateway/myprovider/$1"/>
>
> the above woks too, but i am tied to the from_user i have defined.
>
> So i wanted to use be able to change the from_user dinamically according
> to my needs at the dialplan level, is that possible ? i have tried the
> following but it doesn't work
>
> <action application="bridge"
> data="{sip_from_user=12345678}sofia/gateway/myprovider/$1"/>
>
> Any suggestions ? or is not possible to define a gateway and change some
> of its parameters in the dialplan ? i just want to use the ping
> functionality and have something more organized.
>
> Slds.
>
>    



From mario_fs at mgtech.com  Tue Apr 10 05:23:04 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 9 Apr 2012 18:23:04 -0700
Subject: [Freeswitch-users] Developers is this a bridge change or
	Jira/Bug issue?
In-Reply-To: <CAKzWOxWr7dNdCOzqoOQZPR2Yakq12-+oQf8veXqCTALXgcKq0Q@mail.gmail.com>
References: <F660A0B2-57A4-4077-86D7-EE9E54D7B750@mgtech.com>
	<CAKzWOxWr7dNdCOzqoOQZPR2Yakq12-+oQf8veXqCTALXgcKq0Q@mail.gmail.com>
Message-ID: <24B935AE-428C-48CA-9902-DEB2D109FCA8@mgtech.com>

100 Percent consistent. If the callers number is in the bridge the bridge rings once and no wait for timeout. If the callers number is not in the bridge it always works fine. Never any deviation. This is not important to me since the workaround is probably better since we are not calling ourself (but would have like a better if/then/else to do it). I just wanted to pass this along in case the action was unintended and may cause other problems if it's a bug. If it was intended then rather than not work it would be better to simply have FS remove the caller from the bridge. The traces are console debug, it you need more let me know, but again, I am running fine with the changes below to accommodate the newer git. Mario G

Caller number not also part of bridge, works always:
http://pastebin.freeswitch.org/18845

Caller number is also part of bridge, fails always:
http://pastebin.freeswitch.org/18846


On Apr 9, 2012, at 12:20 PM, Michael Collins wrote:

> I would like to see the console debug log and corresponding sip trace on a working call vs. non-working call. Does it only fail if the destination number is the same as the origination_caller_id_number, or are there other scenarios where it works (or doesn't work)? Throw that up on pb and let's have a look.
> 
> -MC
> 
> On Mon, Apr 9, 2012 at 10:39 AM, Mario G <mario_fs at mgtech.com> wrote:
> Testing the recent gits introduced a problem which I was able to work around but wondered if this is an intended change or bug: If the caller_id_number is one of the numbers in a bridge command only 1 ring is done and the bridge ends, it appears as if originate_timeout is ignored, the trace is normal. Prior to the git update the bridge would continue to ring to the timeout and the caller would be bridged to all the numbers in the bridge so any phone could answer the call.
> 
> What used to work in older git:
>             <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/>
> 
> Newer git. Now I have to execute an extension to check who is calling so they are not in the bridge:
>     <extension name="bridge_call">
>       <condition field="destination_number" expression="^bridge_call$">
>       </condition>   
>         <condition field="${caller_id_number_work}" expression="^9161112222$" break="on-true">
>             <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444"/>
>         </condition>
>         <condition field="${caller_id_number_work}" expression="^9163334444$" break="on-true">
>             <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222"/>
>         </condition>
>         <condition field="${caller_id_number_work}" expression="^(\d+)$">
>             <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/>
>         </condition>
>     </extension>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From ocset at the800group.com  Tue Apr 10 09:20:05 2012
From: ocset at the800group.com (ocset)
Date: Tue, 10 Apr 2012 13:20:05 +0800
Subject: [Freeswitch-users] Problem bridging to an external mobile number
Message-ID: <4F83C305.7070606@the800group.com>

Hi

My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines. 
The first line is a POTS line and the second line is a GSM gateway 
device. Both devices work correctly if I dial out but when I try and 
bridge between the two, nothing works as expected. The bridge is from 
the POTS line to a mobile number using the GSM gateway.

I have created the following bridge for the POTS line (phone numbers 
removed for privacy reasons)

<extension name="pots_line">
<condition field="destination_number" expression="^(xxxxxxxxxx)$">
<action application="bridge" 
data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062"/>
</condition>
</extension>

The bridge does eventually happen (long delay) but neither party can 
hear each other. Also, when I hang-up with one or both phones, the 
connection does not terminate and I have to reboot the GXW4104 to break 
the connection.

I have bought the Freeswitch books and this seems to be something that 
should easily work. I have not added any variables into the <extension> 
definition as this is a simple bridge.

What have I missed?

Thanks in advance


From mthakershi at gmail.com  Tue Apr 10 09:22:47 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 10 Apr 2012 00:22:47 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
Message-ID: <CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>

Upgraded FS to latest version before coming back for help.

Now, I get error saying:
[ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
[ERR] switch_core_io.c:1081 Codec G.729 encoder error!

I think what is happening is, it agrees to use G.729 codec but as soon as
TTS is opened, there is an error. I read in documentation that mod_g729 is
for free codec.

Please help.

Here is the debug trace of the call:
----------------------------------

2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=root 29993 29993 IN IP4 IP1
s=session
c=IN IP4 IP1
t=0 0
m=audio 15582 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20


2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[G722
1:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:2
0:0]/[G7221:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[AMR:
96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:2
0:0]/[AMR:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[SPEE
X:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:2
0:0]/[SPEEX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[G729
:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec
sofia/sipinterface_1/Phone2 at x.24
1.99.201 G729/8000 20 ms 160 samples 8000 bits

2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111
sofia/sipinterface_1/Phone2 at x.241.99
.201 Original read codec set to G729:18

2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_NEW

2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380
(sofia/sipinterface_1/Phone2@
IP1) State NEW

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786
(sofia/sipinterface_1/Phone2 at IP1) Stat
e Change CS_NEW -> CS_INIT

2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_INIT

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
(sofia/sipinterface_1/Phone2@
IP1) State INIT

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85
sofia/sipinterface_1/Phone2 at IP1 SOFI
A INIT

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125
(sofia/sipinterface_1/Phone2 at IP1) S
tate Change CS_INIT -> CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
(sofia/sipinterface_1/Phone2@
IP1) State INIT going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886
(sofia/sipinterface_1/Phone2 at x.241.99.
201) Callstate Change DOWN -> RINGING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/Phone2@
IP1) State ROUTING

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
sofia/sipinterface_1/Phone2 at IP1 SOF
IA ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard ROUTING

2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
+1Phone2 <Phone2>->Phone1 in context inbound
Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing [inbound->vitel-inbound]
continue=fa
lse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound]
destination_num
ber(8774542559) =~ // break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action transfer(1056 XML default)

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
(sofia/sipinterface_1/Phone2@
IP1) State Change CS_ROUTING -> CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/Phone2@
IP1) State ROUTING going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/Phone2@
IP1) State EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
sofia/sipinterface_1/Phone2 at IP1 SOF
IA EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)

2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711
(sofia/sipinterface_1/Phone2 at IP1)
 State Change CS_EXECUTE -> CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer
sofia/sipinterface_1/Phone2 at x.24
1.99.201 to XML[1056 at default]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/Phone2@
IP1) State EXECUTE going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/Phone2@
IP1) State ROUTING

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
sofia/sipinterface_1/Phone2 at IP1 SOF
IA ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard ROUTING

2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
+1Phone2 <Phone2>->1056
in context default
Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing [default->CHPhoneAsmtDev]
continue=f
alse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev]
destination_nu
mber(1056) =~ /^105\d$/ break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action managed(clsAsmtApp)

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
(sofia/sipinterface_1/Phone2@
IP1) State Change CS_ROUTING -> CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/Phone2@
IP1) State ROUTING going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/Phone2@
IP1) State EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
sofia/sipinterface_1/Phone2 at IP1 SOF
IA EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
EXECUTE sofia/sipinterface_1/Phone2 at IP1 managed(clsAsmtApp)

2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed:
attempting to run applica
tion 'clsAsmtApp'.

2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172
CoreSession::seHangupHook, hangup_func: 00000
000

2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound
call.

2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number:
Phone2

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP
[sofia/sipinterface_1/Phone2 at x.2
41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20

2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send
payload to 101

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive
payload to 101

2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP
sofia/sipinterface_1/Phone2 at x.241.
99.201:
v=0
o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56
s=FreeSWITCH
c=IN IP4 64.22.232.56
t=0 0
m=audio 16542 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244
(sofia/sipinterface_1/Phone2 at x.241.99.
201) Callstate Change RINGING -> ACTIVE

2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel
[sofia/sipinterface_1/Phone2 at x.2
41.99.201] has been answered

2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/Phone2 at x.241.99.20
1 entering state [completed][200]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to
customer DB.

2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite

2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
Activated

2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking
text: <break strength='medium
'/>Hello.<break strength='medium'/>Welcome

2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/Phone2 at x.241.99.20
1 entering state [ready][200]

2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable
in passthrough mode!

2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder
error!

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done speaking
text

2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227
ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside
clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
Activated

2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking
text: <break strength='medium
'/>We must verify your identity.

2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable
in passthrough mode!

2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder
error!

2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done speaking
text

2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec
Activated L16 at 8000hz 1 channels
20ms

2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable
in passthrough mode!

2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder
error!
----------------------

On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Try absolute_codec_string<http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Tue Apr 10 09:32:01 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 10 Apr 2012 09:32:01 +0400
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <4F83C305.7070606@the800group.com>
References: <4F83C305.7070606@the800group.com>
Message-ID: <CANBhyPC0XE_jy06wwXq7iQE1JfdC0urBwjGmD=0fg4yB15n_nw@mail.gmail.com>

Try give us more debug info. The good start would be
/log 7
sofia profile internal siptrace on
fsctl send_siphup (for rotate freeswitch.log)

Make call, paste freeswitch.log to pastebin.freswitch.org

10 ?????? 2012 ?. 9:20 ???????????? ocset <ocset at the800group.com> ???????:

> Hi
>
> My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
> The first line is a POTS line and the second line is a GSM gateway
> device. Both devices work correctly if I dial out but when I try and
> bridge between the two, nothing works as expected. The bridge is from
> the POTS line to a mobile number using the GSM gateway.
>
> I have created the following bridge for the POTS line (phone numbers
> removed for privacy reasons)
>
> <extension name="pots_line">
> <condition field="destination_number" expression="^(xxxxxxxxxx)$">
> <action application="bridge"
> data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062"/>
> </condition>
> </extension>
>
> The bridge does eventually happen (long delay) but neither party can
> hear each other. Also, when I hang-up with one or both phones, the
> connection does not terminate and I have to reboot the GXW4104 to break
> the connection.
>
> I have bought the Freeswitch books and this seems to be something that
> should easily work. I have not added any variables into the <extension>
> definition as this is a simple bridge.
>
> What have I missed?
>
> Thanks in advance
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From peter.olsson at visionutveckling.se  Tue Apr 10 09:34:12 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 10 Apr 2012 05:34:12 +0000
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>,
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>

TTS needs a real license for G729 - please check here how to purchase it: http://www.freeswitch.org/node/235

The free G729 modules is for passthrough only, if you generate audio on the FS instance, it will require encoder/decoder licenses.

/Peter

________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Malay Thakershi [mthakershi at gmail.com]
Skickat: den 10 april 2012 07:22
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Possible to disable core codecs?

Upgraded FS to latest version before coming back for help.

Now, I get error saying:
[ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
[ERR] switch_core_io.c:1081 Codec G.729 encoder error!

I think what is happening is, it agrees to use G.729 codec but as soon as TTS is opened, there is an error. I read in documentation that mod_g729 is for free codec.

Please help.

Here is the debug trace of the call:
----------------------------------

2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=root 29993 29993 IN IP4 IP1
s=session
c=IN IP4 IP1
t=0 0
m=audio 15582 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20


2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G722
1:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72
21:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2
0:0]/[G7221:115:32000:20:48000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[AMR:
96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[AMR
:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2
0:0]/[AMR:96:8000:20:12200]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[SPEE
X:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[SPE
EX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2
0:0]/[SPEEX:99:32000:20:44000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G729
:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72
9:18:8000:20:8000]

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec sofia/sipinterface_1/Phone2 at x.24
1.99.201 G729/8000 20 ms 160 samples 8000 bits

2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/Phone2 at x.241.99
.201 Original read codec set to G729:18

2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_NEW

2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380 (sofia/sipinterface_1/Phone2@
IP1) State NEW

2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101

2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786 (sofia/sipinterface_1/Phone2 at IP1) Stat
e Change CS_NEW -> CS_INIT

2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_INIT

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@
IP1) State INIT

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85 sofia/sipinterface_1/Phone2 at IP1 SOFI
A INIT

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125 (sofia/sipinterface_1/Phone2 at IP1) S
tate Change CS_INIT -> CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@
IP1) State INIT going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886 (sofia/sipinterface_1/Phone2 at x.241.99.
201) Callstate Change DOWN -> RINGING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@
IP1) State ROUTING

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2 at IP1 SOF
IA ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard ROUTING

2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 <Phone2>->Phone1 in context inbound
Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing [inbound->vitel-inbound] continue=fa
lse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound] destination_num
ber(8774542559) =~ // break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action transfer(1056 XML default)

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@
IP1) State Change CS_ROUTING -> CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@
IP1) State ROUTING going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@
IP1) State EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2 at IP1 SOF
IA EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)

2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711 (sofia/sipinterface_1/Phone2 at IP1)
 State Change CS_EXECUTE -> CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer sofia/sipinterface_1/Phone2 at x.24
1.99.201 to XML[1056 at default]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@
IP1) State EXECUTE going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@
IP1) State ROUTING

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2 at IP1 SOF
IA ROUTING

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard ROUTING

2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 <Phone2>->1056
in context default
Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing [default->CHPhoneAsmtDev] continue=f
alse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev] destination_nu
mber(1056) =~ /^105\d$/ break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action managed(clsAsmtApp)

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@
IP1) State Change CS_ROUTING -> CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278
29132 at IP1 [BREAK]

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@
IP1) State ROUTING going to sleep

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@
IP1) Running State Change CS_EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@
IP1) State EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2 at IP1 SOF
IA EXECUTE

2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
EXECUTE sofia/sipinterface_1/Phone2 at IP1 managed(clsAsmtApp)

2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed: attempting to run applica
tion 'clsAsmtApp'.

2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: 00000
000

2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound call.

2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number: Phone2

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP [sofia/sipinterface_1/Phone2 at x.2
41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20

2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send payload to 101

2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive payload to 101

2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP sofia/sipinterface_1/Phone2 at x.241.
99.201:
v=0
o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56
s=FreeSWITCH
c=IN IP4 64.22.232.56
t=0 0
m=audio 16542 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244 (sofia/sipinterface_1/Phone2 at x.241.99.
201) Callstate Change RINGING -> ACTIVE

2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel [sofia/sipinterface_1/Phone2 at x.2
41.99.201] has been answered

2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2 at x.241.99.20
1 entering state [completed][200]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782
9132 at IP1 [BREAK]

2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to customer DB.

2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite

2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated

2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: <break strength='medium
'/>Hello.<break strength='medium'/>Welcome

2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2 at x.241.99.20
1 entering state [ready][200]

2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!

2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done speaking text

2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227 ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside
clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite

2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated

2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: <break strength='medium
'/>We must verify your identity.

2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!

2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!

2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done speaking text

2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels
20ms

2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!

2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
----------------------

On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
Try absolute_codec_string<http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
-Avi

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


!DSPAM:4f83c26e32762080149657!


From miha at softnet.si  Tue Apr 10 11:20:12 2012
From: miha at softnet.si (Miha)
Date: Tue, 10 Apr 2012 09:20:12 +0200
Subject: [Freeswitch-users] Cookbook example
Message-ID: <4F83DF2C.4020904@softnet.si>

Hi,

I need a little help. I am trying to use example for inserting XML to 
mysql cdrtable. I have add id before caller_id_name as I need Id in 
mysql table. Problem is that I am getting random numbers in ID field, 
sometimes values for ID are the same, so sql perl script return error 
and values are not inserted.

Can you help me how can I modify so that this will work properly.

use strict;
use warnings;
use CGI;
use DBI;
use Data::Dump qw(dump);
$|++;
my $q = CGI->new;
my $raw_cdr = $q->param('cdr');

my @all_fields = qw(id caller_id_name caller_id_number
destination_number context start_stamp answer_stamp end_stamp
duration billsec hangup_cause uuid bleg_uuid
accountcode domain_name);
my @fields;
my @values;
foreach my $field (@all_fields) {
next unless $raw_cdr =~ m/$field>(.*?)</;
push @fields, $field;
push @values, "'" . urldecode($1) . "'";
}
my $cdr_line;
my $query = sprintf(
"INSERT INTO %s (%s) VALUES (%s);",
'radacct_1', join(',', @fields), join(',', @values)
);
my $db = 
DBI->connect('DBI:mysql:dbname=voipbill_slo;host=xxx.xxx.xxx.xxx','xxxxxx', 
'xxxxx');
$db->do($query);
print $q->header();
sub urldecode {
my $url = shift;
$url =~ s/%([a-fA-F0-9]{2,2})/chr(hex($1))/eg;
return $url;
}
[root at fs1 cgi-bin]# cat /var/log/secure

Thank you!

Regards,
Miha
-------------- next part --------------
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From Daniel.Knaggs at realitysolutions.co.uk  Tue Apr 10 11:54:37 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Tue, 10 Apr 2012 07:54:37 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAKzWOxXVEzZcR80qvW1aKOzM-3EY4rBHH8GTLxAorGxFCmmeVQ@mail.gmail.com>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
	<CAKzWOxXVEzZcR80qvW1aKOzM-3EY4rBHH8GTLxAorGxFCmmeVQ@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05BED6@RSV16.realitysolutions.local>

Michael,

1. Forgive my ignorance but why would I open a ticket with Sangoma when my card is made by Digium?  Or have I missed something here?
2. Another user has contacted me directly with the same issue as we're getting, but he's using a Sangoma PRI card - I've asked him to post it on the mailing list.  The only thing is common is that we're both using PRI and of course FreeSwitch.
3. As for my signature, I do find it a bit ridiculous that I'm being discriminated because of this.  However, for a test I will reply to this e-mail in plain text rather than HTML to see what it does.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 04 April 2012 18:35
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls


On Wed, Apr 4, 2012 at 3:25 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk> wrote:
Update, it's happened while tracing on span 1:4 so I've copied the in-s1c4 and out-s1c4 files - trace is still running.  The call was inbound.

Also it appears we're now seeing other DTMF tones in coming in as well (numbers when they haven't been pressed on the remote end)...


Which files do you want and where you want them?


I recommend that you open a ticket w/ Sangoma and ask them to analyze the traces. The critical thing is to find out if these are false positives, phantoms, or stuff actually coming down the line from the carrier. I think Sangoma is in the best position to help.
-MC

P.S. - Regarding the footer - yes, it is annoying, but if you can't turn it off then you can't turn it off. I recommend doing what Patrick suggests and getting a gmail account so that you can be free of the footer spam. If you *really* want to help us out, I'd say investigate the Mailman project and see if it has a feature that can auto-strip the junk off of the end of an email. :)

Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel:    01482 373104
Mobile: 07932 408313
Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
http://





From miha at softnet.si  Tue Apr 10 12:21:42 2012
From: miha at softnet.si (Miha)
Date: Tue, 10 Apr 2012 10:21:42 +0200
Subject: [Freeswitch-users] Cookbook example
In-Reply-To: <4F83DF2C.4020904@softnet.si>
References: <4F83DF2C.4020904@softnet.si>
Message-ID: <4F83ED96.8010609@softnet.si>

On 4/10/2012 9:20 AM, Miha wrote:
> Hi,
>
> I need a little help. I am trying to use example for inserting XML to 
> mysql cdrtable. I have add id before caller_id_name as I need Id in 
> mysql table. Problem is that I am getting random numbers in ID field, 
> sometimes values for ID are the same, so sql perl script return error 
> and values are not inserted.
>
> Can you help me how can I modify so that this will work properly.
>
> use strict;
> use warnings;
> use CGI;
> use DBI;
> use Data::Dump qw(dump);
> $|++;
> my $q = CGI->new;
> my $raw_cdr = $q->param('cdr');
>
> my @all_fields = qw(id caller_id_name caller_id_number
> destination_number context start_stamp answer_stamp end_stamp
> duration billsec hangup_cause uuid bleg_uuid
> accountcode domain_name);
> my @fields;
> my @values;
> foreach my $field (@all_fields) {
> next unless $raw_cdr =~ m/$field>(.*?)</;
> push @fields, $field;
> push @values, "'" . urldecode($1) . "'";
> }
> my $cdr_line;
> my $query = sprintf(
> "INSERT INTO %s (%s) VALUES (%s);",
> 'radacct_1', join(',', @fields), join(',', @values)
> );
> my $db = 
> DBI->connect('DBI:mysql:dbname=voipbill_slo;host=xxx.xxx.xxx.xxx','xxxxxx', 
> 'xxxxx');
> $db->do($query);
> print $q->header();
> sub urldecode {
> my $url = shift;
> $url =~ s/%([a-fA-F0-9]{2,2})/chr(hex($1))/eg;
> return $url;
> }
> [root at fs1 cgi-bin]# cat /var/log/secure
>
> Thank you!
>
> Regards,
> Miha
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
HI,

I was having same problem with auto increment in mysql.

Now it works.

Regards,
Miha
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From roland at haenel.me  Tue Apr 10 12:22:39 2012
From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=)
Date: Tue, 10 Apr 2012 10:22:39 +0200
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
In-Reply-To: <4F7E5B73.5000004@communicatefreely.net>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
	<CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>
	<4F7E5B73.5000004@communicatefreely.net>
Message-ID: <CAETTZf5eF=dy23Pw7uQpxK4BFrwP8F0sHeXGTfzXH1O01LW8Xg@mail.gmail.com>

I found this in mod_conference.c:

/* Set MOH from variable if not set */
if (zstr(conference->moh_sound)) {
conference->moh_sound = switch_core_strdup(conference->pool,
switch_channel_get_variable(channel, "*conference_moh_sound*"));
}

This evaluate channel variable conference_moh_sound only if the
"conference" command creates a new conference. Give it a try. ;-)

Greetings,
Roland


2012/4/6 Tim St. Pierre <fs-list at communicatefreely.net>

> xml_curl is worth the effort, as it lets you dynamically set anything in
> a conference profile - from sample rate, to default energy levels.  If
> you are using a database already, it's not a complicated script to
> write, and it lets you do all of that on the fly.
>
> -Tim
>
> Michael Collins wrote:
> > I have not seen a way to do this dynamically other than to use
> > xml_curl and serve up a conf profile with a custom moh-sound param.
> > -MC
> >
> > On Thu, Apr 5, 2012 at 6:55 AM, Lappe, Adam <Adam.Lappe at qsc.de
> > <mailto:Adam.Lappe at qsc.de>> wrote:
> >
> >     Hi all,
> >
> >
> >
> >     i am trying to dynamically set the mod-sound variable in
> >     mod_conference, so I can play different music on hold
> >
> >     to my conference rooms without having to create multiple profiles.
> >
> >
> >
> >     Is there a way to set my moh-sound path while I create the
> conference?
> >
> >
> >
> >     Or can I change the moh-sound file via the event socket in a
> >     existing conference?
> >
> >     I think this would also lead to success.
> >
> >
> >
> >     Thanks in advance,
> >
> >     Adam
> >
> >
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gru?,
Roland
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From anita.hall at simmortel.com  Tue Apr 10 12:33:39 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Tue, 10 Apr 2012 14:03:39 +0530
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones
 when phone is ringing; possible?
In-Reply-To: <4F830CB0.6020301@fahrwerk-berlin.de>
References: <4F830CB0.6020301@fahrwerk-berlin.de>
Message-ID: <CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>

I think, no. You can play only after having answered the call.

regards,
Anita



On Mon, Apr 9, 2012 at 9:52 PM, Fahrwerk Kurierkollektiv (Technik) <
tech at fahrwerk-berlin.de> wrote:

> Hi all,
>
> Is it possible to use moh and if a phone is ringing to play ring tones
> to the calling party?
>
> Thanks in advance,
> Georg
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From azza.miled at gmail.com  Tue Apr 10 12:35:08 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 10 Apr 2012 09:35:08 +0100
Subject: [Freeswitch-users] IVR
Message-ID: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>

Hi

I want to make an IVR with lua. I am e real begginer and I don't know where
to place my script lua ( the path) to be functionnel. Can any one help
please (any intersting tutoriel can help)
Thanks
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From gcd at i.ph  Tue Apr 10 13:03:12 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 10 Apr 2012 17:03:12 +0800
Subject: [Freeswitch-users] IVR
In-Reply-To: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
References: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
Message-ID: <CAHKqMb5N78QD_CAhSd4ZdbWRAuNTS3qB38VEij82x8cj9ncyUw@mail.gmail.com>

have u read this already?
http://wiki.freeswitch.org/wiki/IVR#Lua_IVRs
-nandy
On Tue, Apr 10, 2012 at 4:35 PM, azza miled <azza.miled at gmail.com> wrote:

> Hi
>
> I want to make an IVR with lua. I am e real begginer and I don't know
> where to place my script lua ( the path) to be functionnel. Can any one
> help please (any intersting tutoriel can help)
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Tue Apr 10 13:03:28 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 10 Apr 2012 13:03:28 +0400
Subject: [Freeswitch-users] IVR
In-Reply-To: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
References: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
Message-ID: <CANBhyPDaFKu5PAarmG4NG=CSgyKH_4Aj-DAkdn4t4uoU_vco+w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example

10 ?????? 2012 ?. 12:35 ???????????? azza miled <azza.miled at gmail.com>???????:

> Hi
>
> I want to make an IVR with lua. I am e real begginer and I don't know
> where to place my script lua ( the path) to be functionnel. Can any one
> help please (any intersting tutoriel can help)
> Thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Adam.Lappe at qsc.de  Tue Apr 10 13:21:18 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Tue, 10 Apr 2012 11:21:18 +0200
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
In-Reply-To: <CAETTZf5eF=dy23Pw7uQpxK4BFrwP8F0sHeXGTfzXH1O01LW8Xg@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
	<CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>
	<4F7E5B73.5000004@communicatefreely.net>
	<CAETTZf5eF=dy23Pw7uQpxK4BFrwP8F0sHeXGTfzXH1O01LW8Xg@mail.gmail.com>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42F@CGNMSG01.oc.qsc.de>

This works perfect.

Thank you.

Regards,
Adam

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Roland H?nel
Gesendet: Dienstag, 10. April 2012 10:23
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] dynamic music on hold for mod_conference

I found this in mod_conference.c:

            /* Set MOH from variable if not set */
            if (zstr(conference->moh_sound)) {
                        conference->moh_sound = switch_core_strdup(conference->pool, switch_channel_get_variable(channel, "conference_moh_sound"));
            }

This evaluate channel variable conference_moh_sound only if the "conference" command creates a new conference. Give it a try. ;-)

Greetings,
Roland


2012/4/6 Tim St. Pierre <fs-list at communicatefreely.net<mailto:fs-list at communicatefreely.net>>
xml_curl is worth the effort, as it lets you dynamically set anything in
a conference profile - from sample rate, to default energy levels.  If
you are using a database already, it's not a complicated script to
write, and it lets you do all of that on the fly.

-Tim

Michael Collins wrote:
> I have not seen a way to do this dynamically other than to use
> xml_curl and serve up a conf profile with a custom moh-sound param.
> -MC
>
> On Thu, Apr 5, 2012 at 6:55 AM, Lappe, Adam <Adam.Lappe at qsc.de<mailto:Adam.Lappe at qsc.de>
> <mailto:Adam.Lappe at qsc.de<mailto:Adam.Lappe at qsc.de>>> wrote:
>
>     Hi all,
>
>
>
>     i am trying to dynamically set the mod-sound variable in
>     mod_conference, so I can play different music on hold
>
>     to my conference rooms without having to create multiple profiles.
>
>
>
>     Is there a way to set my moh-sound path while I create the conference?
>
>
>
>     Or can I change the moh-sound file via the event socket in a
>     existing conference?
>
>     I think this would also lead to success.
>
>
>
>     Thanks in advance,
>
>     Adam
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org<mailto:consulting at freeswitch.org> <mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>     <mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Gru?,
Roland


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From azza.miled at gmail.com  Tue Apr 10 13:36:59 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 10 Apr 2012 10:36:59 +0100
Subject: [Freeswitch-users] IVR
In-Reply-To: <CANBhyPDaFKu5PAarmG4NG=CSgyKH_4Aj-DAkdn4t4uoU_vco+w@mail.gmail.com>
References: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
	<CANBhyPDaFKu5PAarmG4NG=CSgyKH_4Aj-DAkdn4t4uoU_vco+w@mail.gmail.com>
Message-ID: <CAGQdkzT8pwGG4kicW6oy1zrqVScVgU1CTou0j30tZRHMY-ezsA@mail.gmail.com>

thaks but i have already read this, a friend gave me an e-book called
freeswitch 1.0.6, Do you thik it is interesting?

2012/4/10 Anton Kvashenkin <anton.jugatsu at gmail.com>

> http://wiki.freeswitch.org/wiki/Lua_Welcome_IVR_Example
>
> 10 ?????? 2012 ?. 12:35 ???????????? azza miled <azza.miled at gmail.com>???????:
>
>> Hi
>>
>> I want to make an IVR with lua. I am e real begginer and I don't know
>> where to place my script lua ( the path) to be functionnel. Can any one
>> help please (any intersting tutoriel can help)
>> Thanks
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gozdal at gmail.com  Tue Apr 10 15:48:25 2012
From: gozdal at gmail.com (Marcin Gozdalik)
Date: Tue, 10 Apr 2012 13:48:25 +0200
Subject: [Freeswitch-users] Skypopen failurereason 7 vaa_input_status false
Message-ID: <CADu1mROQ9bKNN44tJs4h-nCmuDsctLLTiVobB_PGSsHDjkxA7g@mail.gmail.com>

Hi

I am trying to get mod_skypopen to work with my FS installation.
I used interactive installer to setup skypopen. It mostly works OK, but the
download path is dead now, and you have to get Skype 2.0.0.72 binary from
elsewhere.

When starting up Skype clients I get:

ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi
ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi

I got it up to a point where I have 2 Skypes on one login running.
mod_skypopen loads and connects to those instances with no problem.

mod_skypopen loads correctly and I can see them as connected:

freeswitch at internal> sk list
sk console is NOT yet assigned
F ID        Name        IB (F/T)    OB (F/T)    State    CallFlw        UUID
= ====      ========      =======     =======    ======    ============
======
  1    [skype101]      0/0         1/1    IDLE    IDLE
* 2    [skype102]      0/0         0/0    IDLE    IDLE

Total Interfaces: 2  IB Calls(Failed/Total): 0/0  OB Calls(Failed/Total):
1/1

I can even send Skype chat messages.

When I try to make a voice call I get:

2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:173
[406c21c|1b4c78b] [DEBUG_SKYPE  173  ][skype101       ][SELECTD,IDLE]
READING: |||CALL 33 STATUS UNPLACED|||
2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:718
[406c21c|1b4c78b] [DEBUG_SKYPE  718  ][skype101       ][DIALING,UNPLACD]
skype_call: 33 is now UNPLACED
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
[406c21c|1b4c78b] [DEBUG_SKYPE  173  ][skype101       ][DIALING,UNPLACD]
READING: |||CALL 33 STATUS ROUTING|||
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:713
[406c21c|1b4c78b] [DEBUG_SKYPE  713  ][skype101       ][DIALING,ROUTING]
skype_call: 33 is now ROUTING
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
[406c21c|1b4c78b] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
READING: |||CALL 33 FAILUREREASON 7|||
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:540
[406c21c|1b4c78b] [DEBUG_SKYPE  540  ][skype101       ][DIALING,ROUTING]
Skype FAILED on skype_call 33. Let's wait for the FAILED message.
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
[406c21c|1b4c78b] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
READING: |||CALL 33 VAA_INPUT_STATUS FALSE|||
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
[406c21c|1b4c78b] [DEBUG_SKYPE  173  ][skype101       ][DIALING,ROUTING]
READING: |||CALL 33 STATUS FAILED|||
2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:687
[406c21c|1b4c78b] [DEBUG_SKYPE  687  ][skype101       ][DIALING,FAILED] we
tried to call Skype on skype_call 33 and Skype has now FAILED
2012-04-10 13:41:17.088638 [DEBUG] mod_skypopen.c:1420
[406c21c|1b4c78b] [DEBUG_SKYPE  1420 ][skype101       ][DOWN,FAILED] skype
call ended

When trying to make a call from "normal" Skype client to skypopen I
received message, that the other end uses too old a Skype client and I
can't call him. So I upgraded the Skype binary to 2.2.0.35 but no luck
either.

Skype processes run as root, and I have the skypopen module loaded and
/dev/dsp present:

root at conf1:~# lsmod | grep skyp
skypopen                4735  0

root at conf1:~# ls -al /dev/dsp
crw-r--r-- 1 root root 14, 3 2012-04-06 17:13 /dev/dsp

I am running 64 bit Ubuntu 10.04.1 (all latest security applied). FS is
FreeSWITCH Version 1.0.head (git-bda1a8d 2012-03-13 15-42-29 +0100)

What am I missing?

As a sidenote, does Skype 2.2 support OSS sound at all? If not than
mod_skypopen became one-way only as Skype refuses to make a call to Skype
2.0 Linux client.

-- 
Marcin Gozdalik
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From vallimamod.abdullah at imtelecom.fr  Tue Apr 10 16:29:45 2012
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Tue, 10 Apr 2012 14:29:45 +0200
Subject: [Freeswitch-users] Cookbook example
In-Reply-To: <4F83DF2C.4020904@softnet.si>
References: <4F83DF2C.4020904@softnet.si>
Message-ID: <17B0D388-DC94-4715-BE94-DBD742645C1F@imtelecom.fr>

Hi,

If the ID field is set as primary index with auto increment, you should not set it in the insert query.
For your script, it looks like just removing the "id" entry @all_fields array should solve the issue.

Best Regards,
Vallimamod.

On Apr 10, 2012, at 9:20 AM, Miha wrote:

> Hi,
> 
> I need a little help. I am trying to use example for inserting XML to mysql cdrtable. I have add id before caller_id_name as I need Id in mysql table. Problem is that I am getting random numbers in ID field, sometimes values for ID are the same, so sql perl script return error and values are not inserted.
> 
> Can you help me how can I modify so that this will work properly.
> 
> use strict;
> use warnings;
> use CGI;
> use DBI;
> use Data::Dump qw(dump);
> $|++;
> my $q = CGI->new;
> my $raw_cdr = $q->param('cdr');
> 
> my @all_fields = qw(id caller_id_name caller_id_number
> destination_number context start_stamp answer_stamp end_stamp
> duration billsec hangup_cause uuid bleg_uuid
> accountcode domain_name);
> my @fields;
> my @values;
> foreach my $field (@all_fields) {
> next unless $raw_cdr =~ m/$field>(.*?)</;
> push @fields, $field;
> push @values, "'" . urldecode($1) . "'";
> }
> my $cdr_line;
> my $query = sprintf(
> "INSERT INTO %s (%s) VALUES (%s);",
> 'radacct_1', join(',', @fields), join(',', @values)
> );
> my $db = DBI->connect('DBI:mysql:dbname=voipbill_slo;host=xxx.xxx.xxx.xxx','xxxxxx', 'xxxxx');
> $db->do($query);
> print $q->header();
> sub urldecode {
> my $url = shift;
> $url =~ s/%([a-fA-F0-9]{2,2})/chr(hex($1))/eg;
> return $url;
> }
> [root at fs1 cgi-bin]# cat /var/log/secure
> 
> Thank you!
> 
> Regards,
> Miha
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ali.jawad at splendor.net  Tue Apr 10 18:00:38 2012
From: ali.jawad at splendor.net (Ali Jawad)
Date: Tue, 10 Apr 2012 17:00:38 +0300
Subject: [Freeswitch-users] mod_nibblebill issue - channel is not
	initialized for billing
Message-ID: <CAG2QdgYPYnBgPqji+0qAr-WhWfn=gA2cL97mfwLwDNb0JwKL7w@mail.gmail.com>

Hi
I have a kamailio server that sends calls to FS, calls then get routed to a
VOIP provider. This works just fine, but calls dont get billed and it seems
billing is not initialized for the calls at hand

Below is the output for the FS console for such a call :

freeswitch at fs-100-151> 2012-04-10 13:52:16.102555 [NOTICE]
switch_channel.c:926 New Channel sofia/internal/support1 at Kamailio.IP
[e41ebdb3-eac2-4104-bad2-ed802bb8e26f]
2012-04-10 13:52:16.102555 [INFO] mod_dialplan_xml.c:485 Processing
support1 <support1>->0091111111 in context public
2012-04-10 13:52:16.102555 [NOTICE] switch_ivr.c:1717 Transfer
sofia/internal/support1 at kamailio.ip to XML[009611111111 at default]
2012-04-10 13:52:16.102555 [INFO] mod_dialplan_xml.c:485 Processing
support1 <support1>->009611111111 in context default
2012-04-10 13:52:16.122486 [NOTICE] switch_channel.c:926 New Channel
sofia/external/85719611111111 [92118bd1-cddd-4854-b303-2525212961f6]
2012-04-10 13:52:17.503477 [NOTICE] sofia_glue.c:4040 Pre-Answer
sofia/external/85719611111111!
2012-04-10 13:52:17.503477 [INFO] switch_ivr_originate.c:3215 Sending early
media
2012-04-10 13:52:17.503477 [NOTICE] mod_sofia.c:2585 Pre-Answer
sofia/internal/support1 at kamailio.ip!
2012-04-10 13:52:17.863305 [INFO] switch_rtp.c:3207 Auto Changing port from
ip1:56854 to ip2:56854
2012-04-10 13:52:20.143135 [NOTICE] sofia.c:6214 Channel
[sofia/external/12349611111111] has been answered
2012-04-10 13:52:20.163119 [NOTICE] switch_ivr_bridge.c:417 Channel
[sofia/internal/support1 at Kamailio.IP] has been answered

I did check the active calls and checked billing status it says not
initialized for call, see :

freeswitch at fs-100-151> nibblebill e41ebdb3-eac2-4104-bad2-ed802bb8e26f check


2012-04-10 13:53:44.602951 [INFO] mod_nibblebill.c:759 Can't check -
channel is not initialized for billing!

Unloading and loading mod_nibblebill occurs without any errors, see :

freeswitch at fs-100-151> unload mod_nibblebill

+OK

2012-04-10 13:57:05.043193 [NOTICE] switch_loadable_module.c:910 Deleting
Application 'nibblebill'
2012-04-10 13:57:05.043193 [NOTICE] switch_loadable_module.c:963 Deleting
API Function 'nibblebill'
2012-04-10 13:57:05.043193 [CONSOLE] switch_loadable_module.c:1784
Stopping: mod_nibblebill
2012-04-10 13:57:05.043193 [NOTICE] switch_event.c:1935 Event Binding
deleted for mod_nibblebill:SESSION_HEARTBEAT
2012-04-10 13:57:05.043193 [CONSOLE] switch_loadable_module.c:1804
mod_nibblebill unloaded.
freeswitch at fs-100-151> load mod_nibblebill
2012-04-10 13:57:09.462673 [INFO] mod_nibblebill.c:228 Connected ODBC DSN:
bandwidth.com
2012-04-10 13:57:09.462673 [CONSOLE] switch_loadable_module.c:1318
Successfully Loaded [mod_nibblebill]
2012-04-10 13:57:09.462673 [NOTICE] switch_loadable_module.c:254 Adding
Application 'nibblebill'
2012-04-10 13:57:09.462673 [NOTICE] switch_loadable_module.c:298 Adding API
Function 'nibblebill'

+OK Reloading XML
+OK

freeswitch at fs-100-151> 2012-04-10 13:57:09.462673 [INFO] mod_enum.c:812
ENUM Reloaded
2012-04-10 13:57:09.462673 [INFO] switch_time.c:1128 Timezone reloaded 530
definitions

My dialplan includes the following :

<include>
        <extension name="Dial Out to Custom GW">
                <condition field="destination_number"
expression="(\d{10})$">
                        <action application="set"
data="nibble_account=${accountcode}"/>
                        <action application="set" data="nibble_rate=0.30"/>
                        <action application="bridge"
data="sofia/gateway/custom/1234$1"/>
                </condition>
        </extension>
</include>

Thanks
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From vipkilla at gmail.com  Tue Apr 10 18:17:13 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Tue, 10 Apr 2012 10:17:13 -0400
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones
 when phone is ringing; possible?
In-Reply-To: <CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>
References: <4F830CB0.6020301@fahrwerk-berlin.de>
	<CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>
Message-ID: <CAC-LwPMqFPudn8OT8fby1RUOnyr83YV9je6oPcdFsJUna=3zHg@mail.gmail.com>

Have you tried setting the variable 'cc_moh_override' ?
I believe you can set it like
 <action application="set"
data="cc_moh_override=tone_stream://%(400,200,400,450);%(400,2200,400,450)"/>

On Tue, Apr 10, 2012 at 4:33 AM, Anita Hall <anita.hall at simmortel.com> wrote:
> I think, no. You can play only after having answered the call.
>
> regards,
> Anita
>
>
>
>
> On Mon, Apr 9, 2012 at 9:52 PM, Fahrwerk Kurierkollektiv (Technik)
> <tech at fahrwerk-berlin.de> wrote:
>>
>> Hi all,
>>
>> Is it possible to use moh and if a phone is ringing to play ring tones
>> to the calling party?
>>
>> Thanks in advance,
>> Georg
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From valernur at yahoo.com  Tue Apr 10 10:23:25 2012
From: valernur at yahoo.com (Valer)
Date: Mon, 9 Apr 2012 23:23:25 -0700 (PDT)
Subject: [Freeswitch-users] Acoustic Echo Cancellation on Freeswitch side
In-Reply-To: <CAHCLHGsnF8-r1iUSWjFwBbS=auBrVD4+uazNMoeZrBHL+EHLDg@mail.gmail.com>
References: <CAPsVOswZhi+ZBpGgsVSsnYA9hPCOWxFc6P77cKwXjMg7xukgPg@mail.gmail.com>
	<20120329163729.09b3d0db@anubis.defcon1>
	<1333032538.23470.164.camel@luna.madrid.commsmundi.com>
	<E22DDCC58C0C42A6AB84CF92A07CE29F@freeswitch.org>
	<CAHCLHGsnF8-r1iUSWjFwBbS=auBrVD4+uazNMoeZrBHL+EHLDg@mail.gmail.com>
Message-ID: <1334039005349-7451910.post@n2.nabble.com>

Actually the *PBXMate* product that you mentioned appears in the 
http://wiki.freeswitch.org/wiki/PBXMate-FreeSWITCH-integration Wiki of
FreeSWITCH  therefore it is compatible with FreeSWITCH. After reading the
product's page, it looks to me like you can first test the *Personal
PBXMate* to verify that the 
http://www.solicall.com/products.html#PersonalPBXMate acoustic echo
cancellation  is working in your environment.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Acoustic-Echo-Cancellation-on-Freeswitch-side-tp7417842p7451910.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Tue Apr 10 14:33:34 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 10 Apr 2012 18:33:34 +0800
Subject: [Freeswitch-users] How to configure media bypass when using
	mod_callcenter
Message-ID: <CAOTmbMzG2Gd92iN-yrN4shMLn0aO_md87N2xTG11nLvQ-sdthg@mail.gmail.com>

Hi Maestros,

i'm using mod_callcenter to realize a simple senario, pstn user call to
media gw, then go to queue number 7000 of fs, then fs hunt a idel
agents(registered users) to accpet the call. it works fine.

however, i would like to make the media bypass the fs after the agent
answer the call, i.e. media-gw<->agent, however, from the trace,the media
still go via fs. from the trace, i saw when fs make the call to agent, the
fs initiate a new "INVITE"  with itself IP address as the connect ip
address in SDP, which caused the bypass failed. (in the attahed trace, .159
is the fs, .50 and .117 are two registered users).

i have configure below, and if two agent call for each other directly, the
media did bypass fs.

internal.xml
    <param name="inbound-bypass-media" value="true"/>
    <param name="inbound-proxy-media" value="false"/>

othe configuraton as following,

dial plan:

    <extension name="7000_Queue" >
   <condition field="destination_number" expression="^7000$">
       <action application="answer"/>
       <action application="playback"
data="/usr/local/freeswitch/sounds/custom/8000/transfer_to_agent.wav"/>
       <action application="callcenter" data="7000_Queue at localhost"/>
       <action application="system" data="mkdir -p
$${base_dir}/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/"/>
   </condition>
</extension>

callcenter.conf.xml:

<configuration name="callcenter.conf" description="CallCenter">
    <settings>
        <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
    </settings>

    <queues>
        <queue name="7000_Queue at localhost">
            <param name="strategy" value="round-robin"/>
            <param name="moh-sound" value="$${hold_music}"/>
            <param name="time-base-score" value="queue"/>
            <param name="max-wait-time" value="0"/>
            <param name="max-wait-time-with-no-agent" value="0"/>
            <param name="tier-rules-apply" value="false"/>
            <param name="tier-rule-wait-second" value="300"/>
            <param name="tier-rule-wait-multiply-level" value="true"/>
            <param name="tier-rule-no-agent-no-wait" value="false"/>
            <param name="discard-abandoned-after" value="60"/>
            <param name="abandoned-resume-allowed" value="false"/>
        </queue>
    </queues>

<!-- WARNING: Configuration of XML Agents will be updated into the DB upon
restart. -->
<!-- WARNING: Configuration of XML Tiers will reset the level and position
if those were supplied. -->
<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS
shared DB setup (Not currently supported anyway) -->
    <agents>
                <agent name="agent1001 at localhost" type="callback"
contact="[call_timeout=10]user/1001@$${domain_name}" status="Available" />
                <agent name="agent1002 at localhost" type="callback"
contact="[call_timeout=10]user/1002@$${domain_name}" status="Available" />
                <agent name="agent1003 at localhost" type="callback"
contact="[call_timeout=10]user/1003@$${domain_name}" status="Available" />

                </agents>

<!-- If no level or position is provided, they will default to 1.  You
should do this to keep db value on restart. -->
    <tiers>
        <tier agent="agent1001 at localhost" queue="7000_Queue at localhost"
level="1" position="1"/>
        <tier agent="agent1002 at localhost" queue="7000_Queue at localhost"
level="1" position="1"/>
        <tier agent="agent1003 at localhost" queue="7000_Queue at localhost"
level="1" position="1"/>

    </tiers>

</configuration>



Could you please any one help, thanks a lot!

BR,
Charles
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From gmaruzz at gmail.com  Tue Apr 10 16:35:18 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 10 Apr 2012 14:35:18 +0200
Subject: [Freeswitch-users] Skypopen failurereason 7 vaa_input_status
	false
In-Reply-To: <CADu1mROQ9bKNN44tJs4h-nCmuDsctLLTiVobB_PGSsHDjkxA7g@mail.gmail.com>
References: <CADu1mROQ9bKNN44tJs4h-nCmuDsctLLTiVobB_PGSsHDjkxA7g@mail.gmail.com>
Message-ID: <CALXCt0ofJuQh7d4xVyKm18W0NhT1NMfyK79TPmniK+TPw6=Q_w@mail.gmail.com>

you're using skype for ALSA

you would have had to find the SAME skype client (eg: skype for OSS)!

Now I fixed the install.pl to checkout the new package (they modified
the name), so, please update your sources and execute install.pl
again.

commit 1247a64794928b1ffa1290b1c7e399d5f432cf40
Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date:   Tue Apr 10 13:37:27 2012 +0200

    skypopen: updating link to skype client in install.pl, now is
http://mirrors.kernel.org/archlinux/community/os/i686/skype-oss-2.0.0.72-3-i686.pkg.tar.xz

-giovanni

On Tue, Apr 10, 2012 at 1:48 PM, Marcin Gozdalik <gozdal at gmail.com> wrote:
> Hi
>
> I am trying to get mod_skypopen to work with my FS installation.
> I used interactive installer to setup skypopen. It mostly works OK, but the
> download path is dead now, and you have to get Skype 2.0.0.72 binary from
> elsewhere.
>
> When starting up Skype clients I get:
>
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
> ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM
> cards.pcm.hdmi
>
> I got it up to a point where I have 2 Skypes on one login running.
> mod_skypopen loads and connects to those instances with no problem.
>
> mod_skypopen loads correctly and I can see them as connected:
>
> freeswitch at internal> sk list
> sk console is NOT yet assigned
> F ID??? ??? Name??? ??? IB (F/T)??? OB (F/T)??? State??? CallFlw??? ??? UUID
> = ====??? ? ========? ??? =======???? =======??? ======??? ============
> ======
> ? 1??? [skype101]??? ? 0/0??? ???? 1/1??? IDLE??? IDLE
> * 2??? [skype102]??? ? 0/0??? ???? 0/0??? IDLE??? IDLE
>
> Total Interfaces: 2? IB Calls(Failed/Total): 0/0? OB Calls(Failed/Total):
> 1/1
>
> I can even send Skype chat messages.
>
> When I try to make a voice call I get:
>
> 2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:173
> [406c21c|1b4c78b] [DEBUG_SKYPE? 173? ][skype101?????? ][SELECTD,IDLE]
> READING: |||CALL 33 STATUS UNPLACED|||
> 2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:718
> [406c21c|1b4c78b] [DEBUG_SKYPE? 718? ][skype101?????? ][DIALING,UNPLACD]
> skype_call: 33 is now UNPLACED
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
> [406c21c|1b4c78b] [DEBUG_SKYPE? 173? ][skype101?????? ][DIALING,UNPLACD]
> READING: |||CALL 33 STATUS ROUTING|||
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:713
> [406c21c|1b4c78b] [DEBUG_SKYPE? 713? ][skype101?????? ][DIALING,ROUTING]
> skype_call: 33 is now ROUTING
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
> [406c21c|1b4c78b] [DEBUG_SKYPE? 173? ][skype101?????? ][DIALING,ROUTING]
> READING: |||CALL 33 FAILUREREASON 7|||
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:540
> [406c21c|1b4c78b] [DEBUG_SKYPE? 540? ][skype101?????? ][DIALING,ROUTING]
> Skype FAILED on skype_call 33. Let's wait for the FAILED message.
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
> [406c21c|1b4c78b] [DEBUG_SKYPE? 173? ][skype101?????? ][DIALING,ROUTING]
> READING: |||CALL 33 VAA_INPUT_STATUS FALSE|||
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173
> [406c21c|1b4c78b] [DEBUG_SKYPE? 173? ][skype101?????? ][DIALING,ROUTING]
> READING: |||CALL 33 STATUS FAILED|||
> 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:687
> [406c21c|1b4c78b] [DEBUG_SKYPE? 687? ][skype101?????? ][DIALING,FAILED] we
> tried to call Skype on skype_call 33 and Skype has now FAILED
> 2012-04-10 13:41:17.088638 [DEBUG] mod_skypopen.c:1420
> [406c21c|1b4c78b] [DEBUG_SKYPE? 1420 ][skype101?????? ][DOWN,FAILED] skype
> call ended
>
> When trying to make a call from "normal" Skype client to skypopen I received
> message, that the other end uses too old a Skype client and I can't call
> him. So I upgraded the Skype binary to 2.2.0.35 but no luck either.
>
> Skype processes run as root, and I have the skypopen module loaded and
> /dev/dsp present:
>
> root at conf1:~# lsmod | grep skyp
> skypopen??????????????? 4735? 0
>
> root at conf1:~# ls -al /dev/dsp
> crw-r--r-- 1 root root 14, 3 2012-04-06 17:13 /dev/dsp
>
> I am running 64 bit Ubuntu 10.04.1 (all latest security applied). FS is
> FreeSWITCH Version 1.0.head (git-bda1a8d 2012-03-13 15-42-29 +0100)
>
> What am I missing?
>
> As a sidenote, does Skype 2.2 support OSS sound at all? If not than
> mod_skypopen became one-way only as Skype refuses to make a call to Skype
> 2.0 Linux client.
>
> --
> Marcin Gozdalik
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Tue Apr 10 16:59:22 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 10 Apr 2012 14:59:22 +0200
Subject: [Freeswitch-users] skypopen mod wont let me call out
In-Reply-To: <1333905729799-7447994.post@n2.nabble.com>
References: <1333819719037-7446255.post@n2.nabble.com>
	<CALXCt0p6j1zWSCsR8aj=7PfHgntYfD=LcXNnfrOAx3Bw+-r3Fw@mail.gmail.com>
	<1333905729799-7447994.post@n2.nabble.com>
Message-ID: <CALXCt0o0imO-vhpMNDP_Nt=gvtTdd5mpXweaAG+XQWLv6YPDRg@mail.gmail.com>

On Sun, Apr 8, 2012 at 7:22 PM, millerman33 <millyman12721 at gmail.com> wrote:
> Thank you Giovanni for your response. I have downloaded and build this
> program about four times now. I ran the freeswitch console in debug mode and
> it showed this when I attempted to make a call.

Hello Millerman,

unfortunately the debug lines alone does not give us the info we need.

Please fill a Jira issue with all the information requested here:

http://jira.freeswitch.org


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From ocset at the800group.com  Tue Apr 10 19:19:36 2012
From: ocset at the800group.com (ocset)
Date: Tue, 10 Apr 2012 23:19:36 +0800
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <CANBhyPC0XE_jy06wwXq7iQE1JfdC0urBwjGmD=0fg4yB15n_nw@mail.gmail.com>
References: <4F83C305.7070606@the800group.com>
	<CANBhyPC0XE_jy06wwXq7iQE1JfdC0urBwjGmD=0fg4yB15n_nw@mail.gmail.com>
Message-ID: <4F844F88.4060606@the800group.com>

Anton

I have posted the log as you requested under user name ocset.

Just to add some more info to the story, I have a Netgear voip modem and 
I am using that voip line to call the POTS line and when I tested it 
this time, it also killed my internet connection and I had to reboot the 
modem.

So to clarify the devices involved:  Voip phone (not managed by 
Freeswitch) -> POTS line (GXW4104) -> freeswitch -> GSM gateway(GXW4104) 
-> mobile phone.

Thanks
Regards
O

On 04/10/2012 01:32 PM, Anton Kvashenkin wrote:
> Try give us more debug info. The good start would be
> /log 7
> sofia profile internal siptrace on
> fsctl send_siphup (for rotate freeswitch.log)
>
> Make call, paste freeswitch.log to pastebin.freswitch.org 
> <http://pastebin.freswitch.org>
>
> 10 ?????? 2012 ?. 9:20 ???????????? ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> ???????:
>
>     Hi
>
>     My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
>     The first line is a POTS line and the second line is a GSM gateway
>     device. Both devices work correctly if I dial out but when I try and
>     bridge between the two, nothing works as expected. The bridge is from
>     the POTS line to a mobile number using the GSM gateway.
>
>     I have created the following bridge for the POTS line (phone numbers
>     removed for privacy reasons)
>
>     <extension name="pots_line">
>     <condition field="destination_number" expression="^(xxxxxxxxxx)$">
>     <action application="bridge"
>     data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062
>     <http://zzzzzzzzzz at 192.168.0.160:5062>"/>
>     </condition>
>     </extension>
>
>     The bridge does eventually happen (long delay) but neither party can
>     hear each other. Also, when I hang-up with one or both phones, the
>     connection does not terminate and I have to reboot the GXW4104 to
>     break
>     the connection.
>
>     I have bought the Freeswitch books and this seems to be something that
>     should easily work. I have not added any variables into the
>     <extension>
>     definition as this is a simple bridge.
>
>     What have I missed?
>
>     Thanks in advance
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fieldpeak at gmail.com  Tue Apr 10 19:19:30 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 10 Apr 2012 23:19:30 +0800
Subject: [Freeswitch-users] round-robin in hunt group
Message-ID: <CAOTmbMx_Ouwyw+xqkhMLqFLcj1VWk9Au16BGcJEV8_nMxAw7xg@mail.gmail.com>

Hi guys,

Who can advise me how can i configure a round-robin when hunt a user within
a hunt group, that mean fs will remember the last user which last call
tried within a hunt group, as i know, it only can call always from the
first user for each call, thanks.

-- 
Regards,
Charles
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From azza.miled at gmail.com  Tue Apr 10 19:21:24 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 10 Apr 2012 16:21:24 +0100
Subject: [Freeswitch-users] nesting IVR with LUA
Message-ID: <CAGQdkzRkEEgznxnr6W6sg3ge1L9j7bFGuWThLKjFrDxQ6sAqnA@mail.gmail.com>

Hi!
I want to write a script of a main IVR which calls child IVRs. My script is
a LUA script. Does any one know the LUA syntax calling a child IV. I found
only the xml one (<entry digits="1" action="menu-sub" param="child_ivr"/>)
Can any one help please??
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From gozdal at gmail.com  Tue Apr 10 19:26:42 2012
From: gozdal at gmail.com (Marcin Gozdalik)
Date: Tue, 10 Apr 2012 17:26:42 +0200
Subject: [Freeswitch-users] Skypopen failurereason 7 vaa_input_status
	false
In-Reply-To: <CALXCt0ofJuQh7d4xVyKm18W0NhT1NMfyK79TPmniK+TPw6=Q_w@mail.gmail.com>
References: <CADu1mROQ9bKNN44tJs4h-nCmuDsctLLTiVobB_PGSsHDjkxA7g@mail.gmail.com>
	<CALXCt0ofJuQh7d4xVyKm18W0NhT1NMfyK79TPmniK+TPw6=Q_w@mail.gmail.com>
Message-ID: <CADu1mRO0OC9wE_8N1pgYpZ6Q2-tZwezcB=BhPpjz0cjpgNywTg@mail.gmail.com>

2012/4/10 Giovanni Maruzzelli <gmaruzz at gmail.com>

> you're using skype for ALSA
>
> you would have had to find the SAME skype client (eg: skype for OSS)!
>

Thanks Giovanni, it works like charm now!

I wasn't aware that Skype comes either in ALSA or in OSS flavor, assumed
support for both is compiled in.

-- 
Marcin Gozdalik
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From msc at freeswitch.org  Tue Apr 10 19:54:55 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 08:54:55 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05BED6@RSV16.realitysolutions.local>
References: <F4F8972C59EAC841B78AFF3FE28E5BBE057826@RSV16.realitysolutions.local>
	<CAKV-jFz_jFeeZ=ryV1U89ts9D1SocC2KoizC+twSsE_Egof-Uw@mail.gmail.com>
	<CAMWiMy_3vJiEX3sSss5hV59=O_wOvA8ZGwDnw=k-hi_6uAPrQw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE058476@RSV16.realitysolutions.local>
	<CAKzWOxWriaYWk5fEZuL_2YuF3SyyTJZL9JCuS9Y-8tFGRc1Xpg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0587B3@RSV16.realitysolutions.local>
	<F4F8972C59EAC841B78AFF3FE28E5BBE0588BA@RSV16.realitysolutions.local>
	<CAKzWOxUdRq711EyULKDWQJQ6Oj6dfptn2FYMazK7zApeRZCO3Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059800@RSV16.realitysolutions.local>
	<CAMwZ-tJQ0pRX7Eq3CfSqUoi1F80ApFv7oeVBWmQh40pXVAXRKg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059EDF@RSV16.realitysolutions.local>
	<CAGTsDStBR_QpceQGtStSw=vEjVKKOSUk-sNSu5GXeYgrQjFDDg@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE059FE5@RSV16.realitysolutions.local>
	<CAKzWOxVOn3a2N0b-fOtO4jh1GrbJJ+ho22n7oVS-p1z9NGxP7Q@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A512@RSV16.realitysolutions.local>
	<CAKzWOxVBmZ0foUk0vz19Q5ntYCBnavU-Xq+keq88Jwb2XksYaw@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A54A@RSV16.realitysolutions.local>
	<CAKzWOxXC0dLSWZ+dYCr-O5on=jwF0c61iOymf0L+-0sCF7Jf7A@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05A8E9@RSV16.realitysolutions.local>
	<CAKzWOxXVEzZcR80qvW1aKOzM-3EY4rBHH8GTLxAorGxFCmmeVQ@mail.gmail.com>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05BED6@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxUas7zuSUZKJCOaBHsJn03apmB2EVaRtf+24T6m5g_CXA@mail.gmail.com>

On Tue, Apr 10, 2012 at 12:54 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

> Michael,
>
> 1. Forgive my ignorance but why would I open a ticket with Sangoma when my
> card is made by Digium?  Or have I missed something here?
>
Sangoma wrote and supports mod_freetdm and they have expertise to diagnose
the issue


> 2. Another user has contacted me directly with the same issue as we're
> getting, but he's using a Sangoma PRI card - I've asked him to post it on
> the mailing list.  The only thing is common is that we're both using PRI
> and of course FreeSwitch.
>
This is why I suspect something is amiss in FreeTDM - it seems to be
happening regardless of the hardware mfg.


> 3. As for my signature, I do find it a bit ridiculous that I'm being
> discriminated because of this.  However, for a test I will reply to this
> e-mail in plain text rather than HTML to see what it does.
>
I wouldn't put too much stock in this. The footer is a bit annoying but out
of the thousands of people who read these emails only a few said something
about it. I honestly don't believe that someone is holding a solution
hostage until you find a workaround for the "footer spam." :)

-MC
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From bdfoster at endigotech.com  Tue Apr 10 19:58:33 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 10 Apr 2012 11:58:33 -0400
Subject: [Freeswitch-users] round-robin in hunt group
In-Reply-To: <CAOTmbMx_Ouwyw+xqkhMLqFLcj1VWk9Au16BGcJEV8_nMxAw7xg@mail.gmail.com>
References: <CAOTmbMx_Ouwyw+xqkhMLqFLcj1VWk9Au16BGcJEV8_nMxAw7xg@mail.gmail.com>
Message-ID: <CAKV-jFyV--+sVSiHN1jOCM9SLk7aGeZ90ByVAMhYJ6FbG2hLWw@mail.gmail.com>

Search the wiki for mod_callcenter.

-bdfoster
On Apr 10, 2012 11:20 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:

> Hi guys,
>
> Who can advise me how can i configure a round-robin when hunt a user
> within a hunt group, that mean fs will remember the last user which last
> call tried within a hunt group, as i know, it only can call always from the
> first user for each call, thanks.
>
> --
> Regards,
> Charles
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 10 20:23:00 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 09:23:00 -0700
Subject: [Freeswitch-users] dynamic music on hold for mod_conference
In-Reply-To: <CAETTZf5eF=dy23Pw7uQpxK4BFrwP8F0sHeXGTfzXH1O01LW8Xg@mail.gmail.com>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E42D@CGNMSG01.oc.qsc.de>
	<CAKzWOxVz9=vJ3VgYLZCsMA9XAthpB_=YVzF77pHY=zuzeV5qZw@mail.gmail.com>
	<4F7E5B73.5000004@communicatefreely.net>
	<CAETTZf5eF=dy23Pw7uQpxK4BFrwP8F0sHeXGTfzXH1O01LW8Xg@mail.gmail.com>
Message-ID: <CAKzWOxWFrPm9Y6xoHL=79zMbkMuFRpWF4NpJD2kiEQE9PYA4kA@mail.gmail.com>

2012/4/10 Roland H?nel <roland at haenel.me>

> I found this in mod_conference.c:
>
> /* Set MOH from variable if not set */
> if (zstr(conference->moh_sound)) {
>  conference->moh_sound = switch_core_strdup(conference->pool,
> switch_channel_get_variable(channel, "*conference_moh_sound*"));
>  }
>
> This evaluate channel variable conference_moh_sound only if the
> "conference" command creates a new conference. Give it a try. ;-)
>
> Greetings,
> Roland
>
>
Roland,
Nice detective work! It turns out that my channel variable lister script
missed this one. I will revisit it and try again. In the meantime, thanks
for digging this up.

-MC
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From marketing at cluecon.com  Tue Apr 10 20:53:57 2012
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 10 Apr 2012 09:53:57 -0700
Subject: [Freeswitch-users] ClueCon 2012: Call For Speakers
Message-ID: <CAKzWOxUrYU9BsTiJg0KgADgenHL=DtyGtrBLaXfb-Z5AvAnsqg@mail.gmail.com>

[image: Inline image 2]
ClueCon 2012 - Call For Speakers

ClueCon - the open source telephony conference by developers, for
developers - would like to announce that we are having an open call for
speaking proposals for this year's event. If you have an idea for a
technical presentation for ClueCon 2012 then we would like to hear about.

What makes a great ClueCon presentation? The tech savvy crowd that attends
ClueCon loves *technical* presentations. In general, the more technical the
presentation, the better. If you are thinking about a presentation then
consider these points:

   - ClueCon talks are 30 minutes in length, including Q&A time with the
   audience
   - ClueCon has a special focus on open source VoIP and telephony projects
   like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio
   - Attendees enjoy hearing about projects built with open source tools
   - Highly technical discussions that show the nuts and bolts are
   especially well-liked
   - The audience appreciates seeing and participating in live
   demonstrations

Please send your proposals to marketing at cluecon.com. Be sure to include a
working title, description of the talk, and name of the presenter. Don't
delay! There are a limited number of openings.

ClueCon 2012 Registration Information

ClueCon 2012 registration is now open. Visit the registration
page<http://www.cluecon.com/register?cc12cfs>for details. Be sure to
book your room at the
Wyndham <http://www.cluecon.com/hotels?cc12cfs> and qualify for the $300
discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if
you have any questions about ClueCon 2012. Also, keep in mind that the
FreeSWITCH community has a conference
call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each
Wednesday at 1PM Eastern time. This is a great opportunity to talk
about open source telephony and get to know a number folks who will be at
ClueCon 2012. Stay tuned for more news about ClueCon speakers, sponsors,
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From mthakershi at gmail.com  Tue Apr 10 22:00:40 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 10 Apr 2012 13:00:40 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
Message-ID: <CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>

Are these licenses supported on windows? If not, how can I use G729 on it?

One license is 1 encoder/decoder per channel. So if I make 30 concurrent
calls, do I need 30 licenses?

Is the $10 fee per month or one-time?

If I can't use G729 at all, what could be my options to save bandwidth +
get acceptable audio quality?

Thanks for responses.

Malay

On Tue, Apr 10, 2012 at 12:34 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> TTS needs a real license for G729 - please check here how to purchase it:
> http://www.freeswitch.org/node/235
>
> The free G729 modules is for passthrough only, if you generate audio on
> the FS instance, it will require encoder/decoder licenses.
>
> /Peter
>
> ________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f?r Malay Thakershi [
> mthakershi at gmail.com]
> Skickat: den 10 april 2012 07:22
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Possible to disable core codecs?
>
> Upgraded FS to latest version before coming back for help.
>
> Now, I get error saying:
> [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
> [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
>
> I think what is happening is, it agrees to use G.729 codec but as soon as
> TTS is opened, there is an error. I read in documentation that mod_g729 is
> for free codec.
>
> Please help.
>
> Here is the debug trace of the call:
> ----------------------------------
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:
> v=0
> o=root 29993 29993 IN IP4 IP1
> s=session
> c=IN IP4 IP1
> t=0 0
> m=audio 15582 RTP/AVP 0 8 3 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[G722
> 1:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[G72
> 21:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[G7221:115:32000:20:48000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[AMR:
> 96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[AMR
> :96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[AMR:96:8000:20:12200]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[SPEE
> X:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[SPE
> EX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [telephone-event:101:8000:2
> 0:0]/[SPEEX:99:32000:20:44000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [GSM:3:8000:20:13200]/[G729
> :18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
> [G729:18:8000:20:8000]/[G72
> 9:18:8000:20:8000]
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec
> sofia/sipinterface_1/Phone2 at x.24
> 1.99.201 G729/8000 20 ms 160 samples 8000 bits
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111
> sofia/sipinterface_1/Phone2 at x.241.99
> .201 Original read codec set to G729:18
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_NEW
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380
> (sofia/sipinterface_1/Phone2@
> IP1) State NEW
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
> send/recv payload to 101
>
> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786
> (sofia/sipinterface_1/Phone2 at IP1) Stat
> e Change CS_NEW -> CS_INIT
>
> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
> (sofia/sipinterface_1/Phone2@
> IP1) State INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85
> sofia/sipinterface_1/Phone2 at IP1 SOFI
> A INIT
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125
> (sofia/sipinterface_1/Phone2 at IP1) S
> tate Change CS_INIT -> CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
> (sofia/sipinterface_1/Phone2@
> IP1) State INIT going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886
> (sofia/sipinterface_1/Phone2 at x.241.99.
> 201) Callstate Change DOWN -> RINGING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard ROUTING
>
> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
> +1Phone2 <Phone2>->Phone1 in context inbound
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
> [inbound->vitel-inbound] continue=fa
> lse
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound]
> destination_num
> ber(8774542559) =~ // break=on-false
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action transfer(1056 XML
> default)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
> (sofia/sipinterface_1/Phone2@
> IP1) State Change CS_ROUTING -> CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard EXECUTE
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711
> (sofia/sipinterface_1/Phone2 at IP1)
>  State Change CS_EXECUTE -> CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer
> sofia/sipinterface_1/Phone2 at x.24
> 1.99.201 to XML[1056 at default]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA ROUTING
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard ROUTING
>
> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
> +1Phone2 <Phone2>->1056
> in context default
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
> [default->CHPhoneAsmtDev] continue=f
> alse
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev]
> destination_nu
> mber(1056) =~ /^105\d$/ break=on-false
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action managed(clsAsmtApp)
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
> (sofia/sipinterface_1/Phone2@
> IP1) State Change CS_ROUTING -> CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/sipinterface_1/97278
> 29132 at IP1 [BREAK]
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
> (sofia/sipinterface_1/Phone2@
> IP1) State ROUTING going to sleep
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
> (sofia/sipinterface_1/Phone2@
> IP1) Running State Change CS_EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
> (sofia/sipinterface_1/Phone2@
> IP1) State EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
> sofia/sipinterface_1/Phone2 at IP1 SOF
> IA EXECUTE
>
> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
> sofia/sipinterface_1/Phone2 at 6
> 6.241.99.201 Standard EXECUTE
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
> EXECUTE sofia/sipinterface_1/Phone2 at IP1 managed(clsAsmtApp)
>
> 2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed:
> attempting to run applica
> tion 'clsAsmtApp'.
>
> 2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172
> CoreSession::seHangupHook, hangup_func: 00000
> 000
>
> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound
> call.
>
> 2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number:
> Phone2
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP
> [sofia/sipinterface_1/Phone2 at x.2
> 41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send
> payload to 101
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive
> payload to 101
>
> 2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP
> sofia/sipinterface_1/Phone2 at x.241.
> 99.201:
> v=0
> o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56
> s=FreeSWITCH
> c=IN IP4 64.22.232.56
> t=0 0
> m=audio 16542 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244
> (sofia/sipinterface_1/Phone2 at x.241.99.
> 201) Callstate Change RINGING -> ACTIVE
>
> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel
> [sofia/sipinterface_1/Phone2 at x.2
> 41.99.201] has been answered
>
> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel
> sofia/sipinterface_1/Phone2 at x.241.99.20
> 1 entering state [completed][200]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
> sofia/sipinterface_1/972782
> 9132 at IP1 [BREAK]
>
> 2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to
> customer DB.
>
> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
> flite
>
> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
> Activated
>
> 2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking
> text: <break strength='medium
> '/>Hello.<break strength='medium'/>Welcome
>
> 2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel
> sofia/sipinterface_1/Phone2 at x.241.99.20
> 1 entering state [ready][200]
>
> 2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done
> speaking text
>
> 2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227
> ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside
> clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
> flite
>
> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
> Activated
>
> 2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking
> text: <break strength='medium
> '/>We must verify your identity.
>
> 2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
>
> 2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done
> speaking text
>
> 2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec
> Activated L16 at 8000hz 1 channels
> 20ms
>
> 2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable
> in passthrough mode!
>
> 2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder
> error!
> ----------------------
>
> On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi at avimarcus.net<mailto:
> avi at avimarcus.net>> wrote:
> Try absolute_codec_string<
> http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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>
>
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>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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From bdfoster at endigotech.com  Tue Apr 10 22:16:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 10 Apr 2012 14:16:27 -0400
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
Message-ID: <CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>

I can't comment on the windows comparability, but it's a one time fee and
is only required when you need to transcode (including media bugs for
recording, TTS, voicemail, etc.). If you don't do that on a channel then
that license isn't used.
On Apr 10, 2012 2:02 PM, "Malay Thakershi" <mthakershi at gmail.com> wrote:

> Are these licenses supported on windows? If not, how can I use G729 on it?
>
> One license is 1 encoder/decoder per channel. So if I make 30 concurrent
> calls, do I need 30 licenses?
>
> Is the $10 fee per month or one-time?
>
> If I can't use G729 at all, what could be my options to save bandwidth +
> get acceptable audio quality?
>
> Thanks for responses.
>
> Malay
>
> On Tue, Apr 10, 2012 at 12:34 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> TTS needs a real license for G729 - please check here how to purchase it:
>> http://www.freeswitch.org/node/235
>>
>> The free G729 modules is for passthrough only, if you generate audio on
>> the FS instance, it will require encoder/decoder licenses.
>>
>> /Peter
>>
>> ________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
>> freeswitch-users-bounces at lists.freeswitch.org] f?r Malay Thakershi [
>> mthakershi at gmail.com]
>> Skickat: den 10 april 2012 07:22
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] Possible to disable core codecs?
>>
>> Upgraded FS to latest version before coming back for help.
>>
>> Now, I get error saying:
>> [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
>> [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
>>
>> I think what is happening is, it agrees to use G.729 codec but as soon as
>> TTS is opened, there is an error. I read in documentation that mod_g729 is
>> for free codec.
>>
>> Please help.
>>
>> Here is the debug trace of the call:
>> ----------------------------------
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP:
>> v=0
>> o=root 29993 29993 IN IP4 IP1
>> s=session
>> c=IN IP4 IP1
>> t=0 0
>> m=audio 15582 RTP/AVP 0 8 3 18 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>>
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G72
>> 21:115:32000:20:48000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMA:8:8000:20:64000]/[G72
>> 21:115:32000:20:48000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[G722
>> 1:115:32000:20:48000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [G729:18:8000:20:8000]/[G72
>> 21:115:32000:20:48000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [telephone-event:101:8000:2
>> 0:0]/[G7221:115:32000:20:48000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
>> send/recv payload to 101
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[AMR
>> :96:8000:20:12200]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMA:8:8000:20:64000]/[AMR
>> :96:8000:20:12200]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[AMR:
>> 96:8000:20:12200]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [G729:18:8000:20:8000]/[AMR
>> :96:8000:20:12200]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [telephone-event:101:8000:2
>> 0:0]/[AMR:96:8000:20:12200]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
>> send/recv payload to 101
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[SPE
>> EX:99:32000:20:44000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMA:8:8000:20:64000]/[SPE
>> EX:99:32000:20:44000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[SPEE
>> X:99:32000:20:44000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [G729:18:8000:20:8000]/[SPE
>> EX:99:32000:20:44000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [telephone-event:101:8000:2
>> 0:0]/[SPEEX:99:32000:20:44000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
>> send/recv payload to 101
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[G72
>> 9:18:8000:20:8000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [PCMA:8:8000:20:64000]/[G72
>> 9:18:8000:20:8000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [GSM:3:8000:20:13200]/[G729
>> :18:8000:20:8000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
>> [G729:18:8000:20:8000]/[G72
>> 9:18:8000:20:8000]
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec
>> sofia/sipinterface_1/Phone2 at x.24
>> 1.99.201 G729/8000 20 ms 160 samples 8000 bits
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111
>> sofia/sipinterface_1/Phone2 at x.241.99
>> .201 Original read codec set to G729:18
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_NEW
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380
>> (sofia/sipinterface_1/Phone2@
>> IP1) State NEW
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
>> send/recv payload to 101
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786
>> (sofia/sipinterface_1/Phone2 at IP1) Stat
>> e Change CS_NEW -> CS_INIT
>>
>> 2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/sipinterface_1/97278
>> 29132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_INIT
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
>> (sofia/sipinterface_1/Phone2@
>> IP1) State INIT
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85
>> sofia/sipinterface_1/Phone2 at IP1 SOFI
>> A INIT
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125
>> (sofia/sipinterface_1/Phone2 at IP1) S
>> tate Change CS_INIT -> CS_ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/sipinterface_1/97278
>> 29132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401
>> (sofia/sipinterface_1/Phone2@
>> IP1) State INIT going to sleep
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886
>> (sofia/sipinterface_1/Phone2 at x.241.99.
>> 201) Callstate Change DOWN -> RINGING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
>> (sofia/sipinterface_1/Phone2@
>> IP1) State ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
>> sofia/sipinterface_1/Phone2 at IP1 SOF
>> IA ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
>> sofia/sipinterface_1/Phone2 at 6
>> 6.241.99.201 Standard ROUTING
>>
>> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
>> +1Phone2 <Phone2>->Phone1 in context inbound
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
>> [inbound->vitel-inbound] continue=fa
>> lse
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound]
>> destination_num
>> ber(8774542559) =~ // break=on-false
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action transfer(1056 XML
>> default)
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
>> (sofia/sipinterface_1/Phone2@
>> IP1) State Change CS_ROUTING -> CS_EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/sipinterface_1/97278
>> 29132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
>> (sofia/sipinterface_1/Phone2@
>> IP1) State ROUTING going to sleep
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
>> (sofia/sipinterface_1/Phone2@
>> IP1) State EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
>> sofia/sipinterface_1/Phone2 at IP1 SOF
>> IA EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
>> sofia/sipinterface_1/Phone2 at 6
>> 6.241.99.201 Standard EXECUTE
>> EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711
>> (sofia/sipinterface_1/Phone2 at IP1)
>>  State Change CS_EXECUTE -> CS_ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/sipinterface_1/97278
>> 29132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer
>> sofia/sipinterface_1/Phone2 at x.24
>> 1.99.201 to XML[1056 at default]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
>> (sofia/sipinterface_1/Phone2@
>> IP1) State EXECUTE going to sleep
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
>> (sofia/sipinterface_1/Phone2@
>> IP1) State ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148
>> sofia/sipinterface_1/Phone2 at IP1 SOF
>> IA ROUTING
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
>> sofia/sipinterface_1/Phone2 at 6
>> 6.241.99.201 Standard ROUTING
>>
>> 2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing
>> +1Phone2 <Phone2>->1056
>> in context default
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 parsing
>> [default->CHPhoneAsmtDev] continue=f
>> alse
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev]
>> destination_nu
>> mber(1056) =~ /^105\d$/ break=on-false
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
>> Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action managed(clsAsmtApp)
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154
>> (sofia/sipinterface_1/Phone2@
>> IP1) State Change CS_ROUTING -> CS_EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/sipinterface_1/97278
>> 29132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410
>> (sofia/sipinterface_1/Phone2@
>> IP1) State ROUTING going to sleep
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362
>> (sofia/sipinterface_1/Phone2@
>> IP1) Running State Change CS_EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417
>> (sofia/sipinterface_1/Phone2@
>> IP1) State EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241
>> sofia/sipinterface_1/Phone2 at IP1 SOF
>> IA EXECUTE
>>
>> 2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
>> sofia/sipinterface_1/Phone2 at 6
>> 6.241.99.201 Standard EXECUTE
>> EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
>> EXECUTE sofia/sipinterface_1/Phone2 at IP1 managed(clsAsmtApp)
>>
>> 2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227
>> FreeSWITCH.Managed: attempting to run applica
>> tion 'clsAsmtApp'.
>>
>> 2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172
>> CoreSession::seHangupHook, hangup_func: 00000
>> 000
>>
>> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound
>> call.
>>
>> 2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number:
>> Phone2
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP
>> [sofia/sipinterface_1/Phone2 at x.2
>> 41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send
>> payload to 101
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf
>> receive payload to 101
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP
>> sofia/sipinterface_1/Phone2 at x.241.
>> 99.201:
>> v=0
>> o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56
>> s=FreeSWITCH
>> c=IN IP4 64.22.232.56
>> t=0 0
>> m=audio 16542 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244
>> (sofia/sipinterface_1/Phone2 at x.241.99.
>> 201) Callstate Change RINGING -> ACTIVE
>>
>> 2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel
>> [sofia/sipinterface_1/Phone2 at x.2
>> 41.99.201] has been answered
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel
>> sofia/sipinterface_1/Phone2 at x.241.99.20
>> 1 entering state [completed][200]
>>
>> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/sipinterface_1/972782
>> 9132 at IP1 [BREAK]
>>
>> 2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to
>> customer DB.
>>
>> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
>> flite
>>
>> 2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
>> Activated
>>
>> 2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking
>> text: <break strength='medium
>> '/>Hello.<break strength='medium'/>Welcome
>>
>> 2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel
>> sofia/sipinterface_1/Phone2 at x.241.99.20
>> 1 entering state [ready][200]
>>
>> 2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>>
>> 2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729
>> encoder error!
>>
>> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done
>> speaking text
>>
>> 2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227
>> ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside
>> clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.
>>
>> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS
>> flite
>>
>> 2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
>> Activated
>>
>> 2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking
>> text: <break strength='medium
>> '/>We must verify your identity.
>>
>> 2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>>
>> 2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729
>> encoder error!
>>
>> 2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done
>> speaking text
>>
>> 2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec
>> Activated L16 at 8000hz 1 channels
>> 20ms
>>
>> 2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable
>> in passthrough mode!
>>
>> 2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729
>> encoder error!
>> ----------------------
>>
>> On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi at avimarcus.net<mailto:
>> avi at avimarcus.net>> wrote:
>> Try absolute_codec_string<
>> http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
>> -Avi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> !DSPAM:4f83c26e32762080149657!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mthakershi at gmail.com  Tue Apr 10 22:42:13 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 10 Apr 2012 13:42:13 -0500
Subject: [Freeswitch-users] mod_spandsp
Message-ID: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>

Is mod_spandsp only for fax? Does it require special hardware?

Can I use codec it provides for regular calls?

Thanks.

Malay
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From mario_fs at mgtech.com  Tue Apr 10 22:52:57 2012
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 10 Apr 2012 11:52:57 -0700
Subject: [Freeswitch-users] mod_spandsp
In-Reply-To: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>
References: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>
Message-ID: <5603FD6F-BE0A-4208-A597-34926E842828@mgtech.com>

It's used by other things, see the wiki http://wiki.freeswitch.org/wiki/Mod_spandsp . I asked this before and was told it is required, I know a basic FS install requires it.

On Apr 10, 2012, at 11:42 AM, Malay Thakershi wrote:

> Is mod_spandsp only for fax? Does it require special hardware?
> 
> Can I use codec it provides for regular calls?
> 
> Thanks.
> 
> Malay
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Tue Apr 10 23:00:26 2012
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 11 Apr 2012 03:00:26 +0800
Subject: [Freeswitch-users] mod_spandsp
In-Reply-To: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>
References: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>
Message-ID: <4F84834A.8030807@coppice.org>

On 04/11/2012 02:42 AM, Malay Thakershi wrote:
> Is mod_spandsp only for fax? 
No
> Does it require special hardware?
No
>
> Can I use codec it provides for regular calls?
Yes
>
> Thanks.
>
> Malay
>
Regards,
Steve



From msc at freeswitch.org  Tue Apr 10 23:15:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 12:15:18 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
Message-ID: <CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>

Sorry, no g729 licenses on non-Linux plaforms. (Sorry Mac, BSD, Windows
users.)
-MC

On Tue, Apr 10, 2012 at 11:16 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> I can't comment on the windows comparability, but it's a one time fee and
> is only required when you need to transcode (including media bugs for
> recording, TTS, voicemail, etc.). If you don't do that on a channel then
> that license isn't used.
>
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From steveu at coppice.org  Tue Apr 10 23:22:14 2012
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 11 Apr 2012 03:22:14 +0800
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
Message-ID: <4F848866.5010008@coppice.org>

True, but sufficient demand could change that. :-)

Steve

On 04/11/2012 03:15 AM, Michael Collins wrote:
> Sorry, no g729 licenses on non-Linux plaforms. (Sorry Mac, BSD, 
> Windows users.)
> -MC
>
> On Tue, Apr 10, 2012 at 11:16 AM, Brian Foster 
> <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>
>     I can't comment on the windows comparability, but it's a one time
>     fee and is only required when you need to transcode (including
>     media bugs for recording, TTS, voicemail, etc.). If you don't do
>     that on a channel then that license isn't used.
>
>



From gmaruzz at gmail.com  Tue Apr 10 23:28:04 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 10 Apr 2012 21:28:04 +0200
Subject: [Freeswitch-users] gsmopen and USB dongle
Message-ID: <CALXCt0o-ovH9sGw4Y2sCU79t7LpcpO0SHpFVZhuvURnGaETo7g@mail.gmail.com>

Hi FreeSWITCHers,

I'm pleased to tell you I'm about to release an updated version of
mod-gsmopen (cellular network voice endpoint and trunking, and sms
gateway).

Most important feature will be support for the cheap and very reliable
huawei family of usb dongles, eg: Huawei E1550 (no need for soundcard,
no cpu load, native pcm stream).

So, get ready, and start looking on ebay or your retail shop for those
E1550 dongles (roughly $20 one item), and start testing the updated
gsmopen module as soon as is released.

Dozen of concurrent gsm channels (calls) support !

Stay tuned for further announcements in some days.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jkomar at jbox.ca  Tue Apr 10 23:40:57 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Tue, 10 Apr 2012 13:40:57 -0600
Subject: [Freeswitch-users] gsmopen and USB dongle
In-Reply-To: <CALXCt0o-ovH9sGw4Y2sCU79t7LpcpO0SHpFVZhuvURnGaETo7g@mail.gmail.com>
References: <CALXCt0o-ovH9sGw4Y2sCU79t7LpcpO0SHpFVZhuvURnGaETo7g@mail.gmail.com>
Message-ID: <CA+zebiaQxPCjpL_Dj_w4JoXnZcYTO3z_BK36Vga4o+Ww6UvPrw@mail.gmail.com>

On Tue, Apr 10, 2012 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Hi FreeSWITCHers,
>
> I'm pleased to tell you I'm about to release an updated version of
> mod-gsmopen (cellular network voice endpoint and trunking, and sms
> gateway).
>
> Most important feature will be support for the cheap and very reliable
> huawei family of usb dongles, eg: Huawei E1550 (no need for soundcard,
> no cpu load, native pcm stream).
>
> So, get ready, and start looking on ebay or your retail shop for those
> E1550 dongles (roughly $20 one item), and start testing the updated
> gsmopen module as soon as is released.
>
> Dozen of concurrent gsm channels (calls) support !
>
> Stay tuned for further announcements in some days.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>

This is great news. I've been wanting to try mod-gsmopen. I'm looking
forward to the updated release.

Jason


From amilkhanzada at gmail.com  Tue Apr 10 23:54:45 2012
From: amilkhanzada at gmail.com (LearningFS)
Date: Tue, 10 Apr 2012 19:54:45 +0000 (UTC)
Subject: [Freeswitch-users]
	=?utf-8?q?Using_Softphone_on_same_machine_as?=
	=?utf-8?q?=09FreeSwitch=3F?=
References: <33649239.post@talk.nabble.com>
	<CAHKqMb635kBHY4-y-bh=4kYSL7xYmAHq8jqiJBkktwMCAMED7Q@mail.gmail.com>
Message-ID: <loom.20120410T213858-112@post.gmane.org>

Nandy Dagondon <gcd <at> i.ph> writes:

> 
> the x-lite is okay but IMHO add at least 2 real phones (IP or ATA) for 
complete testing e.g. conference calls, call forwarding, etc.
> 

Hi Nandy,

Thanks for the response. I finally got a softphone working by:
1. In "conf/vars.xml", changing 'data="$${local_ip_v4}"' to 
'data="domain=127.0.0.1"'
2. Downloading Twinkle via Ubuntu package manager
3. Changing Twinkle to use Port 5062 in System Settings->Network->Sip Port (this 
was necessary because FS was locking port 5060 as everything is on the same 
machine)
4. Setting up my SIP account in Twinkle using my IP address shown as "inet 
addr:10.0.2.15" after running ifconfig
5. Adding a contact with phone number "sip:1019 at 10.0.2.15"
6. Right clicking this contact and sending an instant message


However, I am getting strange problems now. Twinkle can make phone calls, but it 
crashes when I try to end a phone call.
Twinkle can send text messages and freeswitch shows "[INFO] mod_sms.c:299 
Processing text message 1004->1019 in context default" but my default.xml script 
is not executed (as seen on http://wiki.freeswitch.org/wiki/Mod_sms). I don't 
get any response saying "Hello, you said hello"


Any ideas?
I'll try digging in to this more, but I really think this has something to do 
with me trying to run everything on the same machine. Is there any setting in 
FreeSwitch to allow this?

Thanks,
Amil




From msc at freeswitch.org  Wed Apr 11 00:14:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 13:14:59 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <4F848866.5010008@coppice.org>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
Message-ID: <CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>

On Tue, Apr 10, 2012 at 12:22 PM, Steve Underwood <steveu at coppice.org>wrote:

> True, but sufficient demand could change that. :-)
>
> Steve
>

Good to know! Next question would be: what constitutes "sufficient demand"?
:) In any case, if you are looking for g729 licenses on a platform other
than Linux then let me know off list. We'll start a pool and if enough
people throw their pennies in we'll move forward.

-MC
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From msc at freeswitch.org  Wed Apr 11 00:19:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 13:19:34 -0700
Subject: [Freeswitch-users] mod_spandsp
In-Reply-To: <4F84834A.8030807@coppice.org>
References: <CAKHK4ZMr1A66h1cZUxU8UtWohDJhjNDUgFZ01qmCy92bfpknUw@mail.gmail.com>
	<4F84834A.8030807@coppice.org>
Message-ID: <CAKzWOxVHrm7Y69tDuEWtXV6fVnKqRWjQNWdi6w=K3epf9Z_DGQ@mail.gmail.com>

FYI, Steve has his own website that discusses the ins and outs of SpanDSP:
http://www.soft-switch.org/

It is an awesome DSP library that is not FreeSWITCH-specific. Check it out!

-MC

On Tue, Apr 10, 2012 at 12:00 PM, Steve Underwood <steveu at coppice.org>wrote:

> On 04/11/2012 02:42 AM, Malay Thakershi wrote:
> > Is mod_spandsp only for fax?
> No
> > Does it require special hardware?
> No
> >
> > Can I use codec it provides for regular calls?
> Yes
> >
> > Thanks.
> >
> > Malay
> >
> Regards,
> Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Wed Apr 11 00:21:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 13:21:22 -0700
Subject: [Freeswitch-users] gsmopen and USB dongle
In-Reply-To: <CALXCt0o-ovH9sGw4Y2sCU79t7LpcpO0SHpFVZhuvURnGaETo7g@mail.gmail.com>
References: <CALXCt0o-ovH9sGw4Y2sCU79t7LpcpO0SHpFVZhuvURnGaETo7g@mail.gmail.com>
Message-ID: <CAKzWOxW5_cQX=h-7TQFT-hZ7SWD-iF409g0J1+LuQPz-q_TDBA@mail.gmail.com>

This is excellent news! Giovanni, we'll definitely be having you on the FS
conf call once this is out and people have had a chance to tinker with it.
-MC

On Tue, Apr 10, 2012 at 12:28 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Hi FreeSWITCHers,
>
> I'm pleased to tell you I'm about to release an updated version of
> mod-gsmopen (cellular network voice endpoint and trunking, and sms
> gateway).
>
> Most important feature will be support for the cheap and very reliable
> huawei family of usb dongles, eg: Huawei E1550 (no need for soundcard,
> no cpu load, native pcm stream).
>
> So, get ready, and start looking on ebay or your retail shop for those
> E1550 dongles (roughly $20 one item), and start testing the updated
> gsmopen module as soon as is released.
>
> Dozen of concurrent gsm channels (calls) support !
>
> Stay tuned for further announcements in some days.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Wed Apr 11 00:25:25 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 13:25:25 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Tomorrow, April 11
Message-ID: <CAKzWOxVTsCjozgU=_-UvxOVSFAdxuKgLDg38-WAUku18ziVhOg@mail.gmail.com>

Hello all!

Tomorrow's agenda page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_11

We are going to be talking about some "what if" scenarios. Feel free to
bring your own, but the two specifically that we'll be discussing tomorrow:

* What if I host my FS-related project on GitHub?
* What if I wanted to build a desk phone with cheap ARM hardware?

Please join the conversation and bring your expertise as well as your
questions for discussion.

Talk to you tomorrow!

-Michael
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From tech at fahrwerk-berlin.de  Wed Apr 11 02:20:44 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Wed, 11 Apr 2012 00:20:44 +0200
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones
 when phone is ringing; possible?
In-Reply-To: <CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>
References: <4F830CB0.6020301@fahrwerk-berlin.de>
	<CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>
Message-ID: <4F84B23C.1040601@fahrwerk-berlin.de>

Hi Anita,

On 10.04.2012 10:33, Anita Hall wrote:
> I think, no. You can play only after having answered the call.

The call is answered. If not, the caller wouldn't hear moh.
Question is: If a caller comes out of a queue and the phones are 
ringing, if FS could then play ringtones instead of moh?

Georg


From tech at fahrwerk-berlin.de  Wed Apr 11 02:21:44 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Wed, 11 Apr 2012 00:21:44 +0200
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones
 when phone is ringing; possible?
In-Reply-To: <CAC-LwPMqFPudn8OT8fby1RUOnyr83YV9je6oPcdFsJUna=3zHg@mail.gmail.com>
References: <4F830CB0.6020301@fahrwerk-berlin.de>	<CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>
	<CAC-LwPMqFPudn8OT8fby1RUOnyr83YV9je6oPcdFsJUna=3zHg@mail.gmail.com>
Message-ID: <4F84B278.3070003@fahrwerk-berlin.de>

Hi Vik,

On 10.04.2012 16:17, Vik Killa wrote:
> Have you tried setting the variable 'cc_moh_override' ?
> I believe you can set it like
>   <action application="set"
> data="cc_moh_override=tone_stream://%(400,200,400,450);%(400,2200,400,450)"/>

Thanks for this. No, haven't tried this yet. Will test tomorrow evening, 
and then report back.

Georg


From mthakershi at gmail.com  Wed Apr 11 03:12:57 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 10 Apr 2012 18:12:57 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
Message-ID: <CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>

Thanks for information.

But I am not able to use any other codec than PCMU/PCMA for some reason. I
read related settings and what they mean.

I am using Vitelity for now and they say they do support G722/GSM/G729 on
their end.

In sip_profiles, I have
<param name="inbound-codec-negotiation" value="greedy"/>
(tried scrooge but no success)

In vars.xml, I have
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,GSM,PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,GSM,PCMU,PCMA"/>

In modules.conf.xml, I have
<load module="mod_spandsp"/>

Debug trace from console:
-----
2012-04-10 18:00:17.430961 [DEBUG] sofia.c:7547 IP IP1 Approved by acl
"Malay_ACL[]". Access Granted.
2012-04-10 18:00:17.430961 [NOTICE] switch_channel.c:926 New Channel
sofia/sipinterface_1/7607058888 at IP2 [45fe1639-ead1-4547-b063-e40f38b5fd04]
2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/7607058888 at IP2 entering state [received][100]
2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=root 15681 15681 IN IP4 IP2
s=session
c=IN IP4 IP2
t=0 0
m=audio 12860 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_NEW
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[LPC:7:8000:90:2400]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[LPC:7:8000:90:2400]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[LPC:7:8000:90:2400]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:380
(sofia/sipinterface_1/7607058888 at IP2) State NEW
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[LPC:7:8000:90:2400]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:20:0]/[LPC:7:8000:90:2400]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[GSM:3:8000:20:13200]/[AAL2-G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[G729:18:8000:20:8000]/[AAL2-G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[telephone-event:101:8000:20:0]/[AAL2-G726-16:124:8000:20:16000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4911 Bah HUMBUG! Sticking
with PCMA at 8000h@20i
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:3006 Set Codec
sofia/sipinterface_1/7607058888 at IP2 PCMA/8000 20 ms 160 samples 64000 bits
2012-04-10 18:00:17.430961 [DEBUG] switch_core_codec.c:111
sofia/sipinterface_1/7607058888 at IP2 Original read codec set to PCMA:8
2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf
send/recv payload to 101
2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5786
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_NEW -> CS_INIT
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_INIT
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:401
(sofia/sipinterface_1/7607058888 at IP2) State INIT
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:85
sofia/sipinterface_1/7607058888 at IP2 SOFIA INIT
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:125
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_INIT -> CS_ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:401
(sofia/sipinterface_1/7607058888 at IP2) State INIT going to sleep
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_channel.c:1886
(sofia/sipinterface_1/7607058888 at IP2) Callstate Change DOWN -> RINGING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/7607058888 at IP2) State ROUTING
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:148
sofia/sipinterface_1/7607058888 at IP2 SOFIA ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/7607058888 at IP2 Standard ROUTING
2012-04-10 18:00:17.430961 [INFO] mod_dialplan_xml.c:485 Processing
+17607058888 <7607058888>->DID1 in context inbound
Dialplan: sofia/sipinterface_1/7607058888 at IP2 parsing
[inbound->vitel-inbound] continue=false
Dialplan: sofia/sipinterface_1/7607058888 at IP2 Regex (PASS) [vitel-inbound]
destination_number(DID1) =~ // break=on-false
Dialplan: sofia/sipinterface_1/7607058888 at IP2 Action transfer(1056 XML
default)
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:154
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_ROUTING -> CS_EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/7607058888 at IP2) State ROUTING going to sleep
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/7607058888 at IP2) State EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:241
sofia/sipinterface_1/7607058888 at IP2 SOFIA EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/7607058888 at IP2 Standard EXECUTE
EXECUTE sofia/sipinterface_1/7607058888 at IP2 transfer(1056 XML default)
2012-04-10 18:00:17.430961 [DEBUG] switch_ivr.c:1711
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_EXECUTE -> CS_ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:731 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [NOTICE] switch_ivr.c:1717 Transfer
sofia/sipinterface_1/7607058888 at IP2 to XML[1056 at default]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/7607058888 at IP2) State EXECUTE going to sleep
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/7607058888 at IP2) State ROUTING
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:148
sofia/sipinterface_1/7607058888 at IP2 SOFIA ROUTING
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/7607058888 at IP2 Standard ROUTING
2012-04-10 18:00:17.430961 [INFO] mod_dialplan_xml.c:485 Processing
+17607058888 <7607058888>->1056 in context default
Dialplan: sofia/sipinterface_1/7607058888 at IP2 parsing
[default->CHPhoneAsmtDev] continue=false
Dialplan: sofia/sipinterface_1/7607058888 at IP2 Regex (PASS) [CHPhoneAsmtDev]
destination_number(1056) =~ /^105\d$/ break=on-false
Dialplan: sofia/sipinterface_1/7607058888 at IP2 Action sleep(1000)
Dialplan: sofia/sipinterface_1/7607058888 at IP2 Action managed(clsAsmtApp)
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:154
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_ROUTING -> CS_EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410
(sofia/sipinterface_1/7607058888 at IP2) State ROUTING going to sleep
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/7607058888 at IP2) State EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:241
sofia/sipinterface_1/7607058888 at IP2 SOFIA EXECUTE
2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/7607058888 at IP2 Standard EXECUTE
EXECUTE sofia/sipinterface_1/7607058888 at IP2 sleep(1000)
EXECUTE sofia/sipinterface_1/7607058888 at IP2 managed(clsAsmtApp)
2012-04-10 18:00:18.451403 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed:
attempting to run application 'clsAsmtApp'.
2012-04-10 18:00:18.650609 [DEBUG] switch_cpp.cpp:1172
CoreSession::seHangupHook, hangup_func: 00000000
2012-04-10 18:00:19.491376 [NOTICE] switch_cpp.cpp:1227 Not an outbound
call.
2012-04-10 18:00:19.491376 [INFO] switch_cpp.cpp:1227 caller_id_number:
7607058888
2012-04-10 18:00:19.491376 [DEBUG] sofia_glue.c:3258 AUDIO RTP
[sofia/sipinterface_1/7607058888 at IP2] 10.25.20.202 port 31458 -> IP2 port 12
odec: 8 ms: 20
2012-04-10 18:00:19.491376 [DEBUG] switch_rtp.c:1669 Not using a timer
2012-04-10 18:00:19.510906 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send
payload to 101
2012-04-10 18:00:19.510906 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive
payload to 101
2012-04-10 18:00:19.510906 [DEBUG] mod_sofia.c:754 Local SDP
sofia/sipinterface_1/7607058888 at IP2:
v=0
o=FreeSWITCH 1334067361 1334067362 IN IP4 64.22.232.56
s=FreeSWITCH
c=IN IP4 64.22.232.56
t=0 0
m=audio 31458 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2012-04-10 18:00:19.510906 [DEBUG] switch_core_session.c:731 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:19.510906 [DEBUG] switch_channel.c:3244
(sofia/sipinterface_1/7607058888 at IP2) Callstate Change RINGING -> ACTIVE
2012-04-10 18:00:19.510906 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:19.510906 [NOTICE] switch_cpp.cpp:599 Channel
[sofia/sipinterface_1/7607058888 at IP2] has been answered
2012-04-10 18:00:19.510906 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/7607058888 at IP2 entering state [completed][200]
2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:20.331166 [INFO] switch_cpp.cpp:1227 OK. Connected to
customer DB.
2012-04-10 18:00:20.331166 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite
2012-04-10 18:00:20.331166 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec
Activated
2012-04-10 18:00:20.390732 [DEBUG] switch_ivr_play_say.c:2160 Speaking
text: <break strength='medium'/>Hello.<break strength='medium'/>Welcome.
2012-04-10 18:00:20.411239 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/7607058888 at IP2 entering state [ready][200]
2012-04-10 18:00:20.611421 [DEBUG] switch_rtp.c:3224 Correct ip/port
confirmed.
2012-04-10 18:00:26.071033 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:26.071033 [DEBUG] switch_channel.c:2848
(sofia/sipinterface_1/7607058888 at IP2) Callstate Change ACTIVE -> HANGUP
2012-04-10 18:00:26.071033 [NOTICE] sofia.c:636 Hangup
sofia/sipinterface_1/7607058888 at IP2 [CS_EXECUTE] [NORMAL_CLEARING]
2012-04-10 18:00:26.071033 [DEBUG] switch_channel.c:2871 Send signal
sofia/sipinterface_1/7607058888 at IP2 [KILL]
2012-04-10 18:00:26.071033 [DEBUG] switch_cpp.cpp:1227 AppFunction is in
hangupCallback.
2012-04-10 18:00:26.071033 [NOTICE] switch_cpp.cpp:1227
45fe1639-ead1-4547-b063-e40f38b5fd04-clsAsmtApp hanging up, UUID:
45fe1639-ead1-4547-b063-e40f38b5f
2012-04-10 18:00:26.071033 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:26.071033 [DEBUG] switch_ivr_play_say.c:2354 done speaking
text
2012-04-10 18:00:26.071033 [NOTICE] switch_cpp.cpp:1227
45fe1639-ead1-4547-b063-e40f38b5fd04-Inside clsAsmtApp.Run (args '');
HookState is CS_HANGUP.
2012-04-10 18:00:26.091539 [DEBUG] switch_cpp.cpp:635 CoreSession::hangup
2012-04-10 18:00:26.231179 [INFO] switch_cpp.cpp:1227 Entered 6
SpeakDelayReport in ThCustomerDb database.
2012-04-10 18:00:26.231179 [INFO] switch_cpp.cpp:1227 Entered 2
PathDelayReport in ThCustomerDb database.
2012-04-10 18:00:26.250709 [INFO] switch_cpp.cpp:1227 Entered 4
IVRAsmtStatusRecord in ThCustomerDb database.
2012-04-10 18:00:26.250709 [DEBUG] switch_cpp.cpp:1007
sofia/sipinterface_1/7607058888 at IP2 destroy/unlink session from object
2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:2287
sofia/sipinterface_1/7607058888 at IP2 skip receive message
[APPLICATION_EXEC_COMPLETE]
nel is hungup already)
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:417
(sofia/sipinterface_1/7607058888 at IP2) State EXECUTE going to sleep
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_HANGUP
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:602
(sofia/sipinterface_1/7607058888 at IP2) State HANGUP
2012-04-10 18:00:26.250709 [DEBUG] mod_sofia.c:469 Channel
sofia/sipinterface_1/7607058888 at IP2 hanging up, cause: NORMAL_CLEARING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:47
sofia/sipinterface_1/7607058888 at IP2 Standard HANGUP, cause: NORMAL_CLEARING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:602
(sofia/sipinterface_1/7607058888 at IP2) State HANGUP going to sleep
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:393
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_HANGUP -> CS_REPORTING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:362
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_REPORTING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:662
(sofia/sipinterface_1/7607058888 at IP2) State REPORTING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:79
sofia/sipinterface_1/7607058888 at IP2 Standard REPORTING, cause:
NORMAL_CLEARING
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:662
(sofia/sipinterface_1/7607058888 at IP2) State REPORTING going to sleep
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:387
(sofia/sipinterface_1/7607058888 at IP2) State Change CS_REPORTING ->
CS_DESTROY
2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1182 Send signal
sofia/sipinterface_1/7607058888 at IP2 [BREAK]
2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1382 Session 1
(sofia/sipinterface_1/7607058888 at IP2) Locked, Waiting on external entities
2012-04-10 18:00:26.250709 [NOTICE] switch_core_session.c:1400 Session 1
(sofia/sipinterface_1/7607058888 at IP2) Ended
2012-04-10 18:00:26.250709 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/sipinterface_1/7607058888 at IP2 [CS_DESTROY]
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:491
(sofia/sipinterface_1/7607058888 at IP2) Callstate Change HANGUP -> DOWN
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:494
(sofia/sipinterface_1/7607058888 at IP2) Running State Change CS_DESTROY
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:504
(sofia/sipinterface_1/7607058888 at IP2) State DESTROY
2012-04-10 18:00:26.250709 [DEBUG] mod_sofia.c:374
sofia/sipinterface_1/7607058888 at IP2 SOFIA DESTROY
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:86
sofia/sipinterface_1/7607058888 at IP2 Standard DESTROY
2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:504
(sofia/sipinterface_1/7607058888 at IP2) State DESTROY going to sleep
-----

Thanks for help. Am I doing something wrong here?

Malay

On Tue, Apr 10, 2012 at 3:14 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Apr 10, 2012 at 12:22 PM, Steve Underwood <steveu at coppice.org>wrote:
>
>> True, but sufficient demand could change that. :-)
>>
>> Steve
>>
>
> Good to know! Next question would be: what constitutes "sufficient
> demand"? :) In any case, if you are looking for g729 licenses on a platform
> other than Linux then let me know off list. We'll start a pool and if
> enough people throw their pennies in we'll move forward.
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Wed Apr 11 03:34:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 16:34:23 -0700
Subject: [Freeswitch-users] IVR
In-Reply-To: <CAGQdkzT8pwGG4kicW6oy1zrqVScVgU1CTou0j30tZRHMY-ezsA@mail.gmail.com>
References: <CAGQdkzTubZYYJySvAf-9wDrNHHaSvPsNxXKpoBu9o1P8cgkEyg@mail.gmail.com>
	<CANBhyPDaFKu5PAarmG4NG=CSgyKH_4Aj-DAkdn4t4uoU_vco+w@mail.gmail.com>
	<CAGQdkzT8pwGG4kicW6oy1zrqVScVgU1CTou0j30tZRHMY-ezsA@mail.gmail.com>
Message-ID: <CAKzWOxWx9Scjg-eH7oT1Goc6PbKXH_ZubK3v7b0V+qCcsL+phQ@mail.gmail.com>

2012/4/10 azza miled <azza.miled at gmail.com>

> thaks but i have already read this, a friend gave me an e-book called
> freeswitch 1.0.6, Do you thik it is interesting?
>
> Since I wrote the Lua chapter in that book I'd definitely have to say that
it is very interesting! :)
-MC
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From msc at freeswitch.org  Wed Apr 11 03:35:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 10 Apr 2012 16:35:36 -0700
Subject: [Freeswitch-users] nesting IVR with LUA
In-Reply-To: <CAGQdkzRkEEgznxnr6W6sg3ge1L9j7bFGuWThLKjFrDxQ6sAqnA@mail.gmail.com>
References: <CAGQdkzRkEEgznxnr6W6sg3ge1L9j7bFGuWThLKjFrDxQ6sAqnA@mail.gmail.com>
Message-ID: <CAKzWOxVLkYg1RnnygBLV+KMNV1p_A98aHOWBZdVONcN8NCAx-w@mail.gmail.com>

Could you paste the script that you have written thus far? It may help us
to know what you're doing before we offer any advice.
-MC

On Tue, Apr 10, 2012 at 8:21 AM, azza miled <azza.miled at gmail.com> wrote:

> Hi!
> I want to write a script of a main IVR which calls child IVRs. My script
> is a LUA script. Does any one know the LUA syntax calling a child IV. I
> found only the xml one (<entry digits="1" action="menu-sub"
> param="child_ivr"/>) Can any one help please??
>
>
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From dujinfang at gmail.com  Wed Apr 11 05:43:12 2012
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 11 Apr 2012 09:43:12 +0800
Subject: [Freeswitch-users] Open Bugs on Jira and Call for help
In-Reply-To: <CBA847DA.30E76%krice@freeswitch.org>
References: <CBA847DA.30E76%krice@freeswitch.org>
Message-ID: <FE9A8882503B48028C522DC68D3CAB0A@gmail.com>

I saw Tony committed f209ae0 (http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=f209ae0191bd2ab09fc76fa4dd7795827951feb6) so in the new update of http://jira.freeswitch.org/browse/FS-4088  I included both file updates again. Hope it is right.


On Monday, April 9, 2012 at 9:01 PM, Ken Rice wrote:

> Re: [Freeswitch-users] Open Bugs on Jira and Call for help The stuff in mod_XXX/conf should be exhaustive for the module...
>  
> The stuff in conf/${style}/* should be a working config specific to the example being presented in ${style}
>  
>  
> As far as wiki differening from code, I think we should go with what the code says...   There are a few cases where this might not be true (in the even of say a regression)
>  
> K
>  
>  
> On 4/9/12 7:48 AM, "Seven Du" <dujinfang at gmail.com> wrote:
>  
> >  
> > Thanks Ken.
> >  
> > see http://jira.freeswitch.org/browse/FS-4088
> >  
> > questions:
> >  
> > 1) I see confs both in mod_xxx/conf and conf/vanilla, should I modify both? I might miss the vanilla story, would anyone tell it briefly?
> >  
> > 2) wiki saids http://wiki.freeswitch.org/wiki/Mod_conference
> >  
> > conference-flags Can be any combination of wait-mod and waste-bandwidth. wait-mod  
> >  
> > but I only see  
> >  
> > for (i = 0; i < argc && argv[i]; i++) {
> > if (!strcasecmp(argv[i], "wait-mod")) {
> > *f |= CFLAG_WAIT_MOD;
> > } else if (!strcasecmp(argv[i], "video-floor-only")) {
> > *f |= CFLAG_VID_FLOOR;
> > } else if (!strcasecmp(argv[i], "video-bridge")) {
> > *f |= CFLAG_VIDEO_BRIDGE;
> > }
> >  
> > is wiki wrong or should we patch mod_conference.c to enable CFLAG_WASTE_FLAG ?  I see this flag is not actually used.
> >  
> > 7.
> >    
> >  
> > On Thursday, March 29, 2012 at 12:59 AM, Ken Rice wrote:
> >  
> > >  
> > > Re: [Freeswitch-users] Open Bugs on Jira and Call for help Actually we can opena  tracking bug on Jira or just email me the diffs I can review and commit that way...
> > >  
> > > Chjeck out git-format-patch its a good way to format a patch so you get credit also
> > >  
> > > K
> > >  
> > >  
> > > On 3/28/12 11:41 AM, "Seven Du" <dujinfang at gmail.com> wrote:
> > >  
> > > >  
> > > > Cool. I'll update some of my open bugs on jira. And for documentation help, I suggest also address the indents, especially don't mix spaces and tabs, I guess all like tabs? Also, is there any coordination of which conf files who's working on or should all choose randomly? And show we commit to git or report on jira?
> > > >  
> > > >    
> > > >  
> > > > On Wednesday, March 28, 2012 at 11:05 PM, Ken Rice wrote:
> > > >  
> > > > >  
> > > > > Open Bugs on Jira and Call for help If you have open bugs on Jira, please make sure they are up to date on any all information....  
> > > > >  
> > > > > If you wish to address a specific bug, please visit us on the weekly conference call so we can address them... We?re getting close to time for rolling Release Candidate for 1.2 and we want to address as many of these as possible...
> > > > >  
> > > > > Also, I?m looking for some configuration documentation help. If you?ve pulled git in the past few days you?ll notice that in the module src directories there is now a conf directory, we need some help with making sure all the configuration options are documented and what their default settings are.  The end goal is to get all the options documented in their default states.  
> > > > > What skills do you need? The ability to read a little C, and the ability to use a text editor to edit/create configs...  
> > > > >  
> > > > > K  
> > > > > _________________________________________________________________________
> > > > > Professional FreeSWITCH Consulting Services:
> > > > > consulting at freeswitch.org
> > > > > http://www.freeswitchsolutions.com
> > > > >  
> > > > > 
> > > > > 
> > > > >  
> > > > > Official FreeSWITCH Sites
> > > > > http://www.freeswitch.org
> > > > > http://wiki.freeswitch.org
> > > > > http://www.cluecon.com
> > > > >  
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > > http://www.freeswitch.org
> > > > >       
> > > >   
> > > >   
> > > >   
> > > >  
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org
> > > > http://www.freeswitchsolutions.com
> > > >  
> > > > 
> > > > 
> > > >  
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > >  
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >  
> > > 
> > > 
> > >  
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >  
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >       
> >   
> >   
> >   
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >  
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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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From maximsuzuki at gmail.com  Wed Apr 11 06:03:07 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Tue, 10 Apr 2012 22:03:07 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
	<CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
Message-ID: <000c01cd1787$3e759710$bb60c530$@gmail.com>

Has anyone reverse engineered the gentls_cert script for the generation of
"CLIENT" certs on Windows using openssl?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Sunday, April 08, 2012 11:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] generating TLS client certificates on
Windows

Hi Maxim,
Take a look at the gentls_cert script it has the openssl.cfg and commands
used to generate the certs. As the wiki details getting openssl working in
windows you should be able to replicate it without issue.

~mitch

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From mitch.capper at gmail.com  Wed Apr 11 07:52:28 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 10 Apr 2012 20:52:28 -0700
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <000c01cd1787$3e759710$bb60c530$@gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
	<CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
	<000c01cd1787$3e759710$bb60c530$@gmail.com>
Message-ID: <CALn233pMuFK5jdN=D1g8wGt4n0-0BJj=N5MOAjF--tJGW58jLQ@mail.gmail.com>

It should not involve much reverse engineering as gentls_cert is
pretty simple and the commands are the same in windows as linux. Also
there is nothing magical about the certificates FS uses so googling:
https://encrypted.google.com/search?q=openssl%20windows%20client%20certificate&ie=UTF-8&oe=UTF-8

Should give decent results and guides, however to help I have attached
the openssl config file that gentls_cert produces which should give
you 90%.  A fairly low bounty could probably get a .net openssl
wrapper for certificate generation made also.

Finally keep in mind cygwin with openssl would probably allow gentls
to work out of the box too.

~Mitch
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From mitch.capper at gmail.com  Wed Apr 11 07:54:46 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 10 Apr 2012 20:54:46 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
Message-ID: <CALn233pNaj3ZytK+sRXTErq09BXuyvfqfnkcBHW+iEJT__MdpQ@mail.gmail.com>

I certainly would like to hear how much demand would be required for
legal windows g729 support.  For mobile softphone developers would be
a nice option for people to be able to purchase g729 support:)

~Mitch


From fieldpeak at gmail.com  Wed Apr 11 07:55:37 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 11 Apr 2012 11:55:37 +0800
Subject: [Freeswitch-users] Pass the Regiseration message from SIP UA to
	remote SIP server
Message-ID: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>

Could anyone advise how to configure FS to Pass the Regiseration message
from SIP UA to remote SIP server like a SBC do, and once sip server down,
the FS will keep the registeration list and take over the sip server job,
thanks.

-- 
Regards,
Charles
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From fieldpeak at gmail.com  Wed Apr 11 07:56:44 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 11 Apr 2012 11:56:44 +0800
Subject: [Freeswitch-users] round-robin in hunt group
In-Reply-To: <CAKV-jFyV--+sVSiHN1jOCM9SLk7aGeZ90ByVAMhYJ6FbG2hLWw@mail.gmail.com>
References: <CAOTmbMx_Ouwyw+xqkhMLqFLcj1VWk9Au16BGcJEV8_nMxAw7xg@mail.gmail.com>
	<CAKV-jFyV--+sVSiHN1jOCM9SLk7aGeZ90ByVAMhYJ6FbG2hLWw@mail.gmail.com>
Message-ID: <CAOTmbMzz5B5PqnHM7C4Y3WaWDi_mjh1obXB9ZmGsdRQ5uDB8Mw@mail.gmail.com>

Thanks Brian.

2012/4/10 Brian Foster <bdfoster at endigotech.com>

> Search the wiki for mod_callcenter.
>
> -bdfoster
> On Apr 10, 2012 11:20 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>
>> Hi guys,
>>
>> Who can advise me how can i configure a round-robin when hunt a user
>> within a hunt group, that mean fs will remember the last user which last
>> call tried within a hunt group, as i know, it only can call always from the
>> first user for each call, thanks.
>>
>> --
>> Regards,
>> Charles
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Charles
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From nasida at live.ru  Wed Apr 11 15:58:09 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 11 Apr 2012 15:58:09 +0400
Subject: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
In-Reply-To: <CAKbxfG_YLLVZvxU-wF8qgnAx3AxQHo82ogn+-zH7CW=WreY7dw@mail.gmail.com>
References: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>,
	<CAKbxfG_YLLVZvxU-wF8qgnAx3AxQHo82ogn+-zH7CW=WreY7dw@mail.gmail.com>
Message-ID: <SNT127-W267A5584CA6F119379B404D5350@phx.gbl>


Thanks Anthony but It looks like I have explained not absolutely accurately. I have set dsn odbc for switch.xml ; sofia.conf (I am not sure that it is necessary) ; internal and external sip profiles. All works fine. I see in mysql my registered devices. My question is :
Does FS use mysql directly in this case? or FS still use sqlite but FS does copying from sqlite to mysql ? For check it I try to delete sqlite db file sofia_reg_internal.db . I have lost my registrations after deleting this file. After restarting, FS has created this file again. FS still use sqlite ? Please correct me if  I am mistaken.
Thanks.

Date: Fri, 6 Apr 2012 21:22:39 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.

You also must seperately enable odbc in the mod Sofia cfg.  Look for dsn in all the configs.
On Apr 6, 2012 1:48 PM, "Yuriy Nasida" <nasida at live.ru> wrote:





Hello guys!
I have set odbc mysql for internal profile. Can I remove sofia_reg_internal.db? I try to delete it but have lost my reg. After restart, FS creates this file again. What is core odbc support? FS just copy the data from file  sofia_reg_internal.db  to mysql via odbc ?

Thanks. 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From nasida at live.ru  Wed Apr 11 16:05:09 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 11 Apr 2012 16:05:09 +0400
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
 sofia_reg_internal.db
In-Reply-To: <CAKbxfG8bYVwe+LFkDoa6tWA2KoF3V8vy4b8iE6m6mywWvdGDPw@mail.gmail.com>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>,
	<SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>,
	<SNT127-W65713C002F6E87F9869B13D5320@phx.gbl>,
	<CAKbxfG8bYVwe+LFkDoa6tWA2KoF3V8vy4b8iE6m6mywWvdGDPw@mail.gmail.com>
Message-ID: <SNT127-W64A638FB74DE9049E307D8D5350@phx.gbl>


Thank again. 
I have fixed this issue.I just have replaced sqlite db file sofia_reg_internal.db by same file from other FS and have done flush all regs.(usual flushing didn't help)
It Looks like this db file was damaged not sure how exactly but replacement fixed this issue. 

> Date: Wed, 4 Apr 2012 13:51:02 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database	sofia_reg_internal.db
> 
> I already advised.
> 
> You did the first one and checked your settings.
> Next capture sip traffic
> 
> sofia global siptrace on
> 
> and look at all the registers for anything anomalous
> 
> On Wed, Apr 4, 2012 at 1:26 PM, Yuriy Nasida <nasida at live.ru> wrote:
> > Anthony, thank you for your answer but I think it is not issue with settings
> > of sip devices. Incorrect additional row may be any of my customers. I mean
> > that the additional row do change after   ./fs_cli -x "sofia profile
> > internal flush_inbound_reg"
> >
> > Also I have this problem after moving to : FreeSWITCH Version 1.0.head
> > (git-2ec02e5 2012-03-01 13-22-52 -0600)  + Centos 6.
> >
> > I really don't understand why i have it just after cleaning of all rows...
> >
> > sqlite> select * from sip_registrations where sip_user=' user2 ';
> > | user2 ||||||||||||||||||||
> >
> > Please advice.
> >
> > Thanks.
> >
> > ________________________________
> > From: nasida at live.ru
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Wed, 4 Apr 2012 21:55:56 +0400
> > Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database
> > sofia_reg_internal.db
> >
> >
> > Update.
> >
> > after ./fs_cli -x "sofia profile internal flush_inbound_reg"  I have
> >
> > sqlite> select * from sip_registrations where sip_user=' user2 ';
> > | user2 ||||||||||||||||||||
> >
> > Why?? I must have clear rows!!
> >
> > Next, I will have 2 rows again but with user3 instead of user1...
> >
> > Thanks.
> >
> > ________________________________
> > From: nasida at live.ru
> > To: freeswitch-users at lists.freeswitch.org
> > Date: Wed, 4 Apr 2012 21:20:51 +0400
> > Subject: [Freeswitch-users] incorrect reg rows in sqlite database
> > sofia_reg_internal.db
> >
> > Hello guys.
> >
> > I have issue with incorrect reg rows in sqlite database
> > sofia_reg_internal.db. it looks like that one registrations are crossed with
> > others. so sometimes i have 2 inbound call for 2 different users instead of
> > one. What it can be ? probably it is some bug of new FS version ? Can you
> > give me some advise ?
> >
> > sqlite> select * from sip_registrations where sip_user='user2';
> >
> > 3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||  user1 <sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
> >
> > 3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2
> > <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
> >
> >
> > Why I have first row here?  When I call for user2 I have two call (for user1
> > an user2) instead of one (for user2).
> > I have separate row for user1...
> >
> > Thanks.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> > Communication Server  Official FreeSWITCH Sites
> > http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> > Communication Server  Official FreeSWITCH Sites
> > http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From maximsuzuki at gmail.com  Wed Apr 11 16:28:39 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Wed, 11 Apr 2012 08:28:39 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <CALn233pMuFK5jdN=D1g8wGt4n0-0BJj=N5MOAjF--tJGW58jLQ@mail.gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
	<CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
	<000c01cd1787$3e759710$bb60c530$@gmail.com>
	<CALn233pMuFK5jdN=D1g8wGt4n0-0BJj=N5MOAjF--tJGW58jLQ@mail.gmail.com>
Message-ID: <004201cd17de$a0e1bee0$e2a53ca0$@gmail.com>

Thanks again Mitch.  I'll chase down both options and report back.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Tuesday, April 10, 2012 11:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] generating TLS client certificates on
Windows

It should not involve much reverse engineering as gentls_cert is pretty
simple and the commands are the same in windows as linux. Also there is
nothing magical about the certificates FS uses so googling:
https://encrypted.google.com/search?q=openssl%20windows%20client%20certifica
te&ie=UTF-8&oe=UTF-8

Should give decent results and guides, however to help I have attached the
openssl config file that gentls_cert produces which should give you 90%.  A
fairly low bounty could probably get a .net openssl wrapper for certificate
generation made also.

Finally keep in mind cygwin with openssl would probably allow gentls to work
out of the box too.

~Mitch



From maximsuzuki at gmail.com  Wed Apr 11 16:36:41 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Wed, 11 Apr 2012 08:36:41 -0400
Subject: [Freeswitch-users] Using TLS on Freeswitch with CsipSimple on
	Android
Message-ID: <004401cd17df$c08a08f0$419e1ad0$@gmail.com>

Wondering if anyone has been able to get TLS and SRTP implemented with two
CsipSimple clients and Freeswitch?  Did you install client certs?  How did
you get the two endpoints to communicate (receive incoming calls from each
other)?

 

Currently, I can get two CSipSimple endpoints to register and signal using
TLS (sslv23) but can't get incoming calls on either of them from either of
them.

 

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From david.villasmil.work at gmail.com  Wed Apr 11 16:38:09 2012
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 11 Apr 2012 14:38:09 +0200
Subject: [Freeswitch-users] nesting IVR with LUA
In-Reply-To: <CAKzWOxVLkYg1RnnygBLV+KMNV1p_A98aHOWBZdVONcN8NCAx-w@mail.gmail.com>
References: <CAGQdkzRkEEgznxnr6W6sg3ge1L9j7bFGuWThLKjFrDxQ6sAqnA@mail.gmail.com>
	<CAKzWOxVLkYg1RnnygBLV+KMNV1p_A98aHOWBZdVONcN8NCAx-w@mail.gmail.com>
Message-ID: <CAFGRPVq_raSG8D_y-NtSdByqRmhwuxeCYH=ifRMdXS-co26XZw@mail.gmail.com>

Why don't you just transfer?

On Wed, Apr 11, 2012 at 1:35 AM, Michael Collins <msc at freeswitch.org> wrote:

> Could you paste the script that you have written thus far? It may help us
> to know what you're doing before we offer any advice.
> -MC
>
>
> On Tue, Apr 10, 2012 at 8:21 AM, azza miled <azza.miled at gmail.com> wrote:
>
>> Hi!
>> I want to write a script of a main IVR which calls child IVRs. My script
>> is a LUA script. Does any one know the LUA syntax calling a child IV. I
>> found only the xml one (<entry digits="1" action="menu-sub"
>> param="child_ivr"/>) Can any one help please??
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From maximsuzuki at gmail.com  Wed Apr 11 17:31:25 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Wed, 11 Apr 2012 09:31:25 -0400
Subject: [Freeswitch-users] generating TLS client certificates on Windows
In-Reply-To: <CALn233pMuFK5jdN=D1g8wGt4n0-0BJj=N5MOAjF--tJGW58jLQ@mail.gmail.com>
References: <000001cd1532$a7afd940$f70f8bc0$@gmail.com>
	<1FFF97C269757C458224B7C895F35F1508BCAB@cantor.std.visionutv.se>
	<000001cd1582$b2713db0$1753b910$@gmail.com>
	<CALn233qpb=d5S0AQ8SX2nGpikzto-f=gUDD-rvJkSCCCofvZ0Q@mail.gmail.com>
	<000c01cd1787$3e759710$bb60c530$@gmail.com>
	<CALn233pMuFK5jdN=D1g8wGt4n0-0BJj=N5MOAjF--tJGW58jLQ@mail.gmail.com>
Message-ID: <006901cd17e7$65ec94a0$31c5bde0$@gmail.com>

Cygwin ran the gentls_cert scripts perfectly on Windows 7 using Cygwin
1.7.13-1

Thanks again Mitch!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Tuesday, April 10, 2012 11:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] generating TLS client certificates on
Windows

It should not involve much reverse engineering as gentls_cert is pretty
simple and the commands are the same in windows as linux. Also there is
nothing magical about the certificates FS uses so googling:
https://encrypted.google.com/search?q=openssl%20windows%20client%20certifica
te&ie=UTF-8&oe=UTF-8

Should give decent results and guides, however to help I have attached the
openssl config file that gentls_cert produces which should give you 90%.  A
fairly low bounty could probably get a .net openssl wrapper for certificate
generation made also.

Finally keep in mind cygwin with openssl would probably allow gentls to work
out of the box too.

~Mitch



From mitch.capper at gmail.com  Wed Apr 11 17:59:09 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 11 Apr 2012 06:59:09 -0700
Subject: [Freeswitch-users] Using TLS on Freeswitch with CsipSimple on
	Android
In-Reply-To: <004401cd17df$c08a08f0$419e1ad0$@gmail.com>
References: <004401cd17df$c08a08f0$419e1ad0$@gmail.com>
Message-ID: <CALn233oZ-WmwE8og76bx8G8kUWYQmpA1ZZSABvt3uMzRib6t5A@mail.gmail.com>

Can they make outgoing calls?

~Mitch


From anita.hall at simmortel.com  Wed Apr 11 18:55:45 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Wed, 11 Apr 2012 20:25:45 +0530
Subject: [Freeswitch-users] load mod_lcr error
Message-ID: <CAGKPtBu_sS_Z3FS9DzLmMUhAk2zJqm9-Xw_=uKE-nTYymukXpg@mail.gmail.com>

Hi

I am getting the following error when giving load mod_lcr on the command
line.

~~
2012-04-11 20:10:56.346283 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0
ERROR: [unixODBC][Driver Manager]Data source name not found, and no default
driver specified
2012-04-11 20:10:56.346283 [CRIT] switch_core_sqldb.c:418 Failure!
2012-04-11 20:10:56.346283 [CRIT] mod_lcr.c:1018 Cannot Open ODBC Database!
2012-04-11 20:10:56.346283 [ERR] mod_lcr.c:2036 Unable to load lcr config
file
2012-04-11 20:10:56.346283 [CRIT] switch_loadable_module.c:1290 Error
Loading module /usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**
~~

I followed steps give on
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

I am on Ubuntu 10.04 LTS and I installed unixodbc, unixodbc-dev, libmyodbc
packages by using apt-get. Then I did ./configure
--enable-core-odbc-support , make, make install on freeswitch and the
config.log shows that unixodbc got recognized and compiled.

I configured the ini files and my odbc works with MySQL

# cat /etc/odbc.ini
[freeswitch]
Driver          = /usr/lib64/odbc/libmyodbc.so
SERVER          = localhost
PORT            = 3306
DATABASE        = freeswitch
OPTION          = 3
USER            = root
PASSWORD        = password

# cat /etc/odbcinst.ini
[MySQL]
Description     = MySQL driver
Driver          = /usr/lib64/odbc/libmyodbc.so
Setup           = /usr/lib64/odbc/libodbcmyS.so
UsageCount      = 1
FileUsage       = 1
Threading       = 0



# isql -v freeswitch
+---------------------------------------+
| Connected!                        |
|                                         |
| sql-statement                    |
| help [tablename]                |
| quit                                   |
|                                         |
+---------------------------------------+
SQL>

My conf/autoload_configs/switch.conf.xml has the line

<param name="core-db-dsn" value="dsn:freeswitch:password" />


My conf/autoload_configs/lcr.conf.xml has the lines

<settings>
    <param name="odbc-dsn" value="dsn:freeswitch:password"/>
</settings>


I shutdown and restarted freeswitch after doing all configurations.





regards,
Anita
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From anita.hall at simmortel.com  Wed Apr 11 19:06:28 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Wed, 11 Apr 2012 20:36:28 +0530
Subject: [Freeswitch-users] combining multiple SIP providers
In-Reply-To: <CAGKPtBsmWA9i1LmoF16Fa1AXDXDQZV7h3c-6a7y15T5Nxc+jQw@mail.gmail.com>
References: <CAKV-jFwdC5qCAfRce-uPJ5gDMOCtehEW6xiQwt4SFWgm40t4Uw@mail.gmail.com>
	<CBA86A31.30E89%krice@freeswitch.org>
	<CAKV-jFw_3AcwYCARuKVVNCznZfovr2PEVnU9MCSVwvpee5H3kQ@mail.gmail.com>
	<CAGKPtBsmWA9i1LmoF16Fa1AXDXDQZV7h3c-6a7y15T5Nxc+jQw@mail.gmail.com>
Message-ID: <CAGKPtBvam-QoRS8oZf=2jdvsMgXFX=L7paibZx969a6NKuPQzA@mail.gmail.com>

I am back :)

I tried both mod_distributor and mod_lcr (looks very promosing and I have
some config missing for which I have opened another thread on the list).

mod_distributor compiled and loads fine but the dialstring given on wiki
can only be used from inside a Dialplan and not from the CLI / API

<action application="bridge"
data="sofia/gateway/${distributor(distributor_list)}/${destination_number}"/>

When I try this with the originate command, I get invalid gateway.  It
looks like I have to expand the variable properly but how ?

freeswitch at internal> originate
{origination_caller_id_number=+911234567890}sofia/gateway/${distributor(knowbd3)}/00911234567890
&park()
-ERR INVALID_GATEWAY
2012-04-11 20:29:56.196288 [DEBUG] switch_ivr_originate.c:1884 Parsing
global variables
2012-04-11 20:29:56.196288 [DEBUG] switch_event.c:1522 Parsing variable
[origination_caller_id_number]=[+911234567890]
2012-04-11 20:29:56.196288 [ERR] mod_sofia.c:4390 Invalid Gateway
'%{distributor(knowbd3)}'

# cat conf/autoload_configs/distributor.conf.xml
<configuration name="distributor.conf" description="Distributor
Configuration">
  <lists>
    <list name="knowbd1" total-weight="1">
      <node name="boloh" weight="1"/>
    </list>

    <list name="knowbd2" total-weight="1">
      <node name="voxbeam" weight="1"/>
    </list>

    <list name="knowbd3" total-weight="3">
      <node name="boloh" weight="1"/>
      <node name="voxbeam" weight="2"/>
    </list>
  </lists>
</configuration>






regards,
Anita



On Mon, Apr 9, 2012 at 9:33 PM, Anita Hall <anita.hall at simmortel.com> wrote:

> Thanks everybody! Much appreciated :)
>
> mod_lcr is what I am trying first!
>
> regards,
> Anita
>
>
>
>
> On Mon, Apr 9, 2012 at 9:06 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Agreed. mod_lcr used here w/ 15ish carriers. Seems like no two carriers
>> ever have the same rate on the same route.
>>
>> -BDF
>>
>>
>> On Mon, Apr 9, 2012 at 11:28 AM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Theres that or mod_lcr or a number of ways to do it...
>>>
>>> But if you are using > 1 carrier if they have varying rates per prefix
>>> you probably want to go ahead and look at LCR so you can least cost
>>> route... Depending on volume this may save you some money
>>>
>>> K
>>>
>>>
>>>
>>> On 4/9/12 10:25 AM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>>>
>>> Is mod_distributor still being developed? That'll take care of you
>>> randomly routing between the two (or more) gateways.
>>>
>>> -BDF
>>>
>>> On Mon, Apr 9, 2012 at 10:57 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>> As Mitch stated, opensips isn't necessary.
>>> -Avi
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From khorsmann at gmail.com  Wed Apr 11 17:42:47 2012
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Wed, 11 Apr 2012 15:42:47 +0200
Subject: [Freeswitch-users] external sip profile - registered users get
	Originate Failed. Cause: USER_NOT_REGISTERED
Message-ID: <CAFArqsaUS_kQNNwL9LHRk9Ztq6vJ-b96W3qiHaQeGD==i3nGxA@mail.gmail.com>

Hi all,


i try to install FS as SBC between public internet and private subnet.
(SIP-User)--->[public-ip]FreeSWITCH[private-ip]--->(IVR System)

If i try to reach an user, i got USER_NOT_REGISTERED. If i registered
to the internal sip profile with the private address, everything works fine.

Any hints?


freeswitch at kamailio1> sofia status profile external
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  212.xx.xx.xx
SIP-IP                  212.xx.xx.xx
URL                     sip:mod_sofia at 212.xx.xx.xx:5060
BIND-URL                sip:mod_sofia at 212.xx.xx.xx:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,GSM
CODECS OUT              PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                4
FAILED-CALLS-IN         2
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           2


freeswitch at kamailio1> sofia status profile internal
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              212.59.42.189,172.20.100.74
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  172.20.100.74
SIP-IP                  172.20.100.74
URL                     sip:mod_sofia at 172.20.100.74:5060
BIND-URL                sip:mod_sofia at 172.20.100.74:5060
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               PCMU,PCMA,GSM
CODECS OUT              PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               1
FAILED-CALLS-OUT        1
REGISTRATIONS           0

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*


From ocset at the800group.com  Wed Apr 11 20:03:00 2012
From: ocset at the800group.com (ocset)
Date: Thu, 12 Apr 2012 00:03:00 +0800
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <CANBhyPC0XE_jy06wwXq7iQE1JfdC0urBwjGmD=0fg4yB15n_nw@mail.gmail.com>
References: <4F83C305.7070606@the800group.com>
	<CANBhyPC0XE_jy06wwXq7iQE1JfdC0urBwjGmD=0fg4yB15n_nw@mail.gmail.com>
Message-ID: <4F85AB34.9090708@the800group.com>

Hi Anton

Here is the URL for the pb - http://pastebin.freeswitch.org/18848

Thanks in advance
O.


On 04/10/2012 01:32 PM, Anton Kvashenkin wrote:
> Try give us more debug info. The good start would be
> /log 7
> sofia profile internal siptrace on
> fsctl send_siphup (for rotate freeswitch.log)
>
> Make call, paste freeswitch.log to pastebin.freswitch.org 
> <http://pastebin.freswitch.org>
>
> 10 ?????? 2012 ?. 9:20 ???????????? ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> ???????:
>
>     Hi
>
>     My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
>     The first line is a POTS line and the second line is a GSM gateway
>     device. Both devices work correctly if I dial out but when I try and
>     bridge between the two, nothing works as expected. The bridge is from
>     the POTS line to a mobile number using the GSM gateway.
>
>     I have created the following bridge for the POTS line (phone numbers
>     removed for privacy reasons)
>
>     <extension name="pots_line">
>     <condition field="destination_number" expression="^(xxxxxxxxxx)$">
>     <action application="bridge"
>     data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062
>     <http://zzzzzzzzzz at 192.168.0.160:5062>"/>
>     </condition>
>     </extension>
>
>     The bridge does eventually happen (long delay) but neither party can
>     hear each other. Also, when I hang-up with one or both phones, the
>     connection does not terminate and I have to reboot the GXW4104 to
>     break
>     the connection.
>
>     I have bought the Freeswitch books and this seems to be something that
>     should easily work. I have not added any variables into the
>     <extension>
>     definition as this is a simple bridge.
>
>     What have I missed?
>
>     Thanks in advance
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mthakershi at gmail.com  Wed Apr 11 20:49:30 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 11 Apr 2012 11:49:30 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CALn233pNaj3ZytK+sRXTErq09BXuyvfqfnkcBHW+iEJT__MdpQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<CALn233pNaj3ZytK+sRXTErq09BXuyvfqfnkcBHW+iEJT__MdpQ@mail.gmail.com>
Message-ID: <CAKHK4ZO03c-pK+1ZC1fe483arVavSAvTdTBbbHpdeNB2uSdJgw@mail.gmail.com>

Thanks for response but I would really like some help on why I am not able
to use any other codec than PCMU/PCMA?

And, if I only use WAV files from my call flow (StreamFile), do I need G729
professional license?

Thanks.

Malay

On Tue, Apr 10, 2012 at 10:54 PM, Mitch Capper <mitch.capper at gmail.com>wrote:

> I certainly would like to hear how much demand would be required for
> legal windows g729 support.  For mobile softphone developers would be
> a nice option for people to be able to purchase g729 support:)
>
> ~Mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jalsot at gmail.com  Wed Apr 11 21:00:40 2012
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Wed, 11 Apr 2012 19:00:40 +0200
Subject: [Freeswitch-users] enable_file_write_buffering
Message-ID: <CAHCLHGt8dGH9dxkdGbM-gK77SUD16B7XJ5mpMo1QfyeWct8EAQ@mail.gmail.com>

Hello,

Does anybody know how much buffering does the
enable_file_write_buffering=true provide?
What I've found from source code that there is a buffer with 65536 length
but not sure how to calculate the actual buffer length in milliseconds - to
estimate IOPS at the end.

I would happily add the info to wiki afterwards.

Kind regards,
   Jalsot
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From maximsuzuki at gmail.com  Wed Apr 11 21:39:27 2012
From: maximsuzuki at gmail.com (Maxim Suzuki)
Date: Wed, 11 Apr 2012 13:39:27 -0400
Subject: [Freeswitch-users] Using TLS on Freeswitch with CsipSimple on
	Android
In-Reply-To: <CALn233oZ-WmwE8og76bx8G8kUWYQmpA1ZZSABvt3uMzRib6t5A@mail.gmail.com>
References: <004401cd17df$c08a08f0$419e1ad0$@gmail.com>
	<CALn233oZ-WmwE8og76bx8G8kUWYQmpA1ZZSABvt3uMzRib6t5A@mail.gmail.com>
Message-ID: <00b501cd180a$0bdcc610$23965230$@gmail.com>

At the moment, they can make outgoing calls to voicemail and to extensions
that are not using TLS.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Wednesday, April 11, 2012 9:59 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Using TLS on Freeswitch with CsipSimple on
Android

Can they make outgoing calls?

~Mitch

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From paul at cupis.co.uk  Wed Apr 11 21:53:40 2012
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 11 Apr 2012 18:53:40 +0100
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
Message-ID: <4F85C524.9070106@cupis.co.uk>

On 11/04/12 00:12, Malay Thakershi wrote:
> Thanks for information.
>
> But I am not able to use any other codec than PCMU/PCMA for some reason.
> I read related settings and what they mean.
>
> I am using Vitelity for now and they say they do support G722/GSM/G729
> on their end.

The far-end SDP says it supports PCMU/PCMA/GSM/G729 only, no G722.

> In vars.xml, I have
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,GSM,PCMU,PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,GSM,PCMU,PCMA"/>

The only matching codecs are PCMA/PCMU/GSM from what you've shown us. 
You have some G726 comparisons in your log as well, but you've not 
mentioned G726 at all.

> 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5572 Remote SDP:
> v=0
> o=root 15681 15681 IN IP4 IP2
> s=session
> c=IN IP4 IP2
> t=0 0
> m=audio 12860 RTP/AVP 0 8 3 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> Audio Codec Compare [G729:18:8000:20:8000]/[LPC:7:8000:90:2400]
> Audio Codec Compare [telephone-event:101:8000:20:0]/[LPC:7:8000:90:2400]
> Set 2833 dtmf send/recv payload to 101
> Audio Codec Compare [PCMU:0:8000:20:64000]/[G726-16:124:8000:20:16000]
> Audio Codec Compare [PCMA:8:8000:20:64000]/[G726-16:124:8000:20:16000]
> Audio Codec Compare [GSM:3:8000:20:13200]/[G726-16:124:8000:20:16000]
> Audio Codec Compare [G729:18:8000:20:8000]/[G726-16:124:8000:20:16000]
> Audio Codec Compare [telephone-event:101:8000:20:0]/[G726-16:124:8000:20:16000]
> Set 2833 dtmf send/recv payload to 101
> Audio Codec Compare [PCMU:0:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
> Audio Codec Compare [PCMA:8:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
> Audio Codec Compare [GSM:3:8000:20:13200]/[AAL2-G726-16:124:8000:20:16000]
> Audio Codec Compare [G729:18:8000:20:8000]/[AAL2-G726-16:124:8000:20:16000]
> Audio Codec Compare [telephone-event:101:8000:20:0]/[AAL2-G726-16:124:8000:20:16000]
> Set 2833 dtmf send/recv payload to 101
> Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> Bah HUMBUG! Sticking with PCMA at 8000h@20i


From anthony.minessale at gmail.com  Wed Apr 11 22:16:53 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 11 Apr 2012 13:16:53 -0500
Subject: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
In-Reply-To: <SNT127-W267A5584CA6F119379B404D5350@phx.gbl>
References: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>
	<CAKbxfG_YLLVZvxU-wF8qgnAx3AxQHo82ogn+-zH7CW=WreY7dw@mail.gmail.com>
	<SNT127-W267A5584CA6F119379B404D5350@phx.gbl>
Message-ID: <CAKbxfG_SmQZXmQuk8iT6zazSJ=Yuh9O3zr8bQaqGXM3omHj1Mg@mail.gmail.com>

if you internal profile xml has this line it will not use sqlite

<param name="odbc-dsn" value="your:dsn"/>

I tried this and then deleted the sqlite file.
The file remained gone and the registrations persisted.
I restarted FS and the file was still not there.
I removed the config option and restarted and the file returned.

Clearly you have some sort of misconfiguration? Which version of FS
are you using?



On Wed, Apr 11, 2012 at 6:58 AM, Yuriy Nasida <nasida at live.ru> wrote:
> Thanks Anthony but It looks like I have explained not absolutely
> accurately.
> I have set dsn odbc for switch.xml ; sofia.conf (I am not sure that it is
> necessary) ; internal and external sip profiles. All works fine. I see in
> mysql my registered devices.
> My question is :
>
> Does FS use mysql directly in this case? or FS still use sqlite but FS does
> copying from sqlite to mysql ?
> For check it I try to delete sqlite db file sofia_reg_internal.db . I have
> lost my registrations after deleting this file. After restarting, FS has
> created this file again. FS still use sqlite ? Please correct me if ?I am
> mistaken.
>
> Thanks.
>
> ________________________________
> Date: Fri, 6 Apr 2012 21:22:39 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
>
>
> You also must seperately enable odbc in the mod Sofia cfg.? Look for dsn in
> all the configs.
> On Apr 6, 2012 1:48 PM, "Yuriy Nasida" <nasida at live.ru> wrote:
>
> Hello guys!
>
> I have set odbc mysql for internal profile. Can I remove
> sofia_reg_internal.db? I try to delete it but have lost my reg. After
> restart, FS creates this file again. What is core odbc support? FS just copy
> the data from file ?sofia_reg_internal.db ?to mysql via odbc ?
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mthakershi at gmail.com  Wed Apr 11 22:31:45 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 11 Apr 2012 13:31:45 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <4F85C524.9070106@cupis.co.uk>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
Message-ID: <CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>

Understood your point about G722. But if my list has GSM at higher priority
over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using
GSM?

I don't know where G726 comparisons are coming from. If I haven't mentioned
it in vars.xml and far-end SDP doesn't support it then why would it even
consider that codec?

Thanks.

Malay


On Wed, Apr 11, 2012 at 12:53 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 11/04/12 00:12, Malay Thakershi wrote:
> > Thanks for information.
> >
> > But I am not able to use any other codec than PCMU/PCMA for some reason.
> > I read related settings and what they mean.
> >
> > I am using Vitelity for now and they say they do support G722/GSM/G729
> > on their end.
>
> The far-end SDP says it supports PCMU/PCMA/GSM/G729 only, no G722.
>
> > In vars.xml, I have
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,GSM,PCMU,PCMA"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,GSM,PCMU,PCMA"/>
>
> The only matching codecs are PCMA/PCMU/GSM from what you've shown us.
> You have some G726 comparisons in your log as well, but you've not
> mentioned G726 at all.
>
> > 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5572 Remote SDP:
> > v=0
> > o=root 15681 15681 IN IP4 IP2
> > s=session
> > c=IN IP4 IP2
> > t=0 0
> > m=audio 12860 RTP/AVP 0 8 3 18 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:3 GSM/8000
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=silenceSupp:off - - - -
> > a=ptime:20
> >
> > Audio Codec Compare [G729:18:8000:20:8000]/[LPC:7:8000:90:2400]
> > Audio Codec Compare [telephone-event:101:8000:20:0]/[LPC:7:8000:90:2400]
> > Set 2833 dtmf send/recv payload to 101
> > Audio Codec Compare [PCMU:0:8000:20:64000]/[G726-16:124:8000:20:16000]
> > Audio Codec Compare [PCMA:8:8000:20:64000]/[G726-16:124:8000:20:16000]
> > Audio Codec Compare [GSM:3:8000:20:13200]/[G726-16:124:8000:20:16000]
> > Audio Codec Compare [G729:18:8000:20:8000]/[G726-16:124:8000:20:16000]
> > Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G726-16:124:8000:20:16000]
> > Set 2833 dtmf send/recv payload to 101
> > Audio Codec Compare
> [PCMU:0:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
> > Audio Codec Compare
> [PCMA:8:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000]
> > Audio Codec Compare
> [GSM:3:8000:20:13200]/[AAL2-G726-16:124:8000:20:16000]
> > Audio Codec Compare
> [G729:18:8000:20:8000]/[AAL2-G726-16:124:8000:20:16000]
> > Audio Codec Compare
> [telephone-event:101:8000:20:0]/[AAL2-G726-16:124:8000:20:16000]
> > Set 2833 dtmf send/recv payload to 101
> > Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000]
> > Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> > Bah HUMBUG! Sticking with PCMA at 8000h@20i
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lazyvirus at gmx.com  Wed Apr 11 22:43:31 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 11 Apr 2012 20:43:31 +0200
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
Message-ID: <20120411204331.0e8017fa@anubis.defcon1>

On Wed, 11 Apr 2012 13:31:45 -0500
Malay Thakershi <mthakershi at gmail.com> wrote:

> Understood your point about G722. But if my list has GSM at higher priority
> over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using
> GSM?
> 
> I don't know where G726 comparisons are coming from. If I haven't mentioned
> it in vars.xml and far-end SDP doesn't support it then why would it even
> consider that codec?

From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
(inbound-codec-negotiation): 
'generous' permits the remote codec list have precedence and 'win'
   the codec negotiation and selection process

JY
-- 
Don't drop acid -- take it pass/fail.
		-- Seen in a Ladies' Room at Harvard


From bdfoster at endigotech.com  Wed Apr 11 23:43:41 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 11 Apr 2012 15:43:41 -0400
Subject: [Freeswitch-users] Pass the Regiseration message from SIP UA to
 remote SIP server
In-Reply-To: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>
References: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>
Message-ID: <CAKV-jFyVsBRB3uKTTSME7FKiGnJrEvTj5AKAyy5uo+FLxcXACQ@mail.gmail.com>

You're going to have to be more clear about your setup and what exactly you
are trying to accomplish.

-BDF
On Apr 10, 2012 11:56 PM, "fieldpeak" <fieldpeak at gmail.com> wrote:

> Could anyone advise how to configure FS to Pass the Regiseration message
> from SIP UA to remote SIP server like a SBC do, and once sip server down,
> the FS will keep the registeration list and take over the sip server job,
> thanks.
>
> --
> Regards,
> Charles
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From nasida at live.ru  Wed Apr 11 23:44:07 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 11 Apr 2012 23:44:07 +0400
Subject: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
In-Reply-To: <CAKbxfG_SmQZXmQuk8iT6zazSJ=Yuh9O3zr8bQaqGXM3omHj1Mg@mail.gmail.com>
References: <SNT127-W16C243A53ADA05A60DF8AED5300@phx.gbl>,
	<CAKbxfG_YLLVZvxU-wF8qgnAx3AxQHo82ogn+-zH7CW=WreY7dw@mail.gmail.com>,
	<SNT127-W267A5584CA6F119379B404D5350@phx.gbl>,
	<CAKbxfG_SmQZXmQuk8iT6zazSJ=Yuh9O3zr8bQaqGXM3omHj1Mg@mail.gmail.com>
Message-ID: <SNT127-W36DAFB96FB71B53695F3F8D5350@phx.gbl>


Anthony, my internal profile really has this line and I see registrations in mysql.Probably I have something wrong but I have not idea what exactly. 
freeswitch at internal> version FreeSWITCH Version 1.0.head (git-2ec02e5 2012-03-01 13-22-52 -0600)
Thanks.

> Date: Wed, 11 Apr 2012 13:16:53 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
> 
> if you internal profile xml has this line it will not use sqlite
> 
> <param name="odbc-dsn" value="your:dsn"/>
> 
> I tried this and then deleted the sqlite file.
> The file remained gone and the registrations persisted.
> I restarted FS and the file was still not there.
> I removed the config option and restarted and the file returned.
> 
> Clearly you have some sort of misconfiguration? Which version of FS
> are you using?
> 
> 
> 
> On Wed, Apr 11, 2012 at 6:58 AM, Yuriy Nasida <nasida at live.ru> wrote:
> > Thanks Anthony but It looks like I have explained not absolutely
> > accurately.
> > I have set dsn odbc for switch.xml ; sofia.conf (I am not sure that it is
> > necessary) ; internal and external sip profiles. All works fine. I see in
> > mysql my registered devices.
> > My question is :
> >
> > Does FS use mysql directly in this case? or FS still use sqlite but FS does
> > copying from sqlite to mysql ?
> > For check it I try to delete sqlite db file sofia_reg_internal.db . I have
> > lost my registrations after deleting this file. After restarting, FS has
> > created this file again. FS still use sqlite ? Please correct me if  I am
> > mistaken.
> >
> > Thanks.
> >
> > ________________________________
> > Date: Fri, 6 Apr 2012 21:22:39 -0500
> > From: anthony.minessale at gmail.com
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] sofia_reg_internal.db and odbc mysql.
> >
> >
> > You also must seperately enable odbc in the mod Sofia cfg.  Look for dsn in
> > all the configs.
> > On Apr 6, 2012 1:48 PM, "Yuriy Nasida" <nasida at live.ru> wrote:
> >
> > Hello guys!
> >
> > I have set odbc mysql for internal profile. Can I remove
> > sofia_reg_internal.db? I try to delete it but have lost my reg. After
> > restart, FS creates this file again. What is core odbc support? FS just copy
> > the data from file  sofia_reg_internal.db  to mysql via odbc ?
> >
> > Thanks.
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> > Communication Server  Official FreeSWITCH Sites
> > http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From govoiper at gmail.com  Wed Apr 11 23:57:13 2012
From: govoiper at gmail.com (SamyGo)
Date: Thu, 12 Apr 2012 00:57:13 +0500
Subject: [Freeswitch-users] Pass the Regiseration message from SIP UA to
 remote SIP server
In-Reply-To: <CAKV-jFyVsBRB3uKTTSME7FKiGnJrEvTj5AKAyy5uo+FLxcXACQ@mail.gmail.com>
References: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>
	<CAKV-jFyVsBRB3uKTTSME7FKiGnJrEvTj5AKAyy5uo+FLxcXACQ@mail.gmail.com>
Message-ID: <CAJUJwtgvK6mrnw-mGs9fMipOxNN1sQrwiatK=kMU0AtkrEV94Q@mail.gmail.com>

Hello,
What i've understood is that you want the REGISTER requests to be relayed
to some other server via FreeSWITCH ! I'd say use OpenSIPS or Kamailio (SIP
proxies) to do this kind of magic.

Regards,
Sammy.


On Thu, Apr 12, 2012 at 12:43 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> You're going to have to be more clear about your setup and what exactly
> you are trying to accomplish.
>
> -BDF
> On Apr 10, 2012 11:56 PM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>
>> Could anyone advise how to configure FS to Pass the Regiseration message
>> from SIP UA to remote SIP server like a SBC do, and once sip server down,
>> the FS will keep the registeration list and take over the sip server job,
>> thanks.
>>
>> --
>> Regards,
>> Charles
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mthakershi at gmail.com  Thu Apr 12 00:48:26 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 11 Apr 2012 15:48:26 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <20120411204331.0e8017fa@anubis.defcon1>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
Message-ID: <CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>

Under my sipinterface_1 profile that I use for these calls, I have:

<param name="inbound-codec-negotiation" value="greedy"/>

On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:

> On Wed, 11 Apr 2012 13:31:45 -0500
> Malay Thakershi <mthakershi at gmail.com> wrote:
>
> > Understood your point about G722. But if my list has GSM at higher
> priority
> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using
> > GSM?
> >
> > I don't know where G726 comparisons are coming from. If I haven't
> mentioned
> > it in vars.xml and far-end SDP doesn't support it then why would it even
> > consider that codec?
>
> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
> (inbound-codec-negotiation):
> 'generous' permits the remote codec list have precedence and 'win'
>   the codec negotiation and selection process
>
> JY
> --
> Don't drop acid -- take it pass/fail.
>                -- Seen in a Ladies' Room at Harvard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From cesar.bermudez at gmail.com  Thu Apr 12 00:53:14 2012
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Wed, 11 Apr 2012 17:53:14 -0300
Subject: [Freeswitch-users]  Mach a pattern to route a group of dids.
Message-ID: <CADrcvV-0i4_nq682UpmfS+kGziTFchRPuL+G-=n7XUv3PbqcZA@mail.gmail.com>

Hi FS'users

I have a block of a did numbers.
 example: 1111 0000 to 1111 2000
i want to route from 1111 0000 to 1111 0099 to another server.
and from 1111 0100 to 1111 2000 to another server.
how can do that in the dialplan?

best regards.
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From tknchris at gmail.com  Thu Apr 12 01:01:28 2012
From: tknchris at gmail.com (chris)
Date: Wed, 11 Apr 2012 17:01:28 -0400
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <4F83C305.7070606@the800group.com>
References: <4F83C305.7070606@the800group.com>
Message-ID: <CAKnNFz-sF2xT5HgRROVG6eiYGUwqGaK_PK5r2q8FvyRDV9kiuw@mail.gmail.com>

what gsm gateway are you using? ive been looking for a good one

On Tue, Apr 10, 2012 at 1:20 AM, ocset <ocset at the800group.com> wrote:

> Hi
>
> My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
> The first line is a POTS line and the second line is a GSM gateway
> device. Both devices work correctly if I dial out but when I try and
> bridge between the two, nothing works as expected. The bridge is from
> the POTS line to a mobile number using the GSM gateway.
>
> I have created the following bridge for the POTS line (phone numbers
> removed for privacy reasons)
>
> <extension name="pots_line">
> <condition field="destination_number" expression="^(xxxxxxxxxx)$">
> <action application="bridge"
> data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062"/>
> </condition>
> </extension>
>
> The bridge does eventually happen (long delay) but neither party can
> hear each other. Also, when I hang-up with one or both phones, the
> connection does not terminate and I have to reboot the GXW4104 to break
> the connection.
>
> I have bought the Freeswitch books and this seems to be something that
> should easily work. I have not added any variables into the <extension>
> definition as this is a simple bridge.
>
> What have I missed?
>
> Thanks in advance
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vetali100 at gmail.com  Thu Apr 12 01:12:43 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 11 Apr 2012 14:12:43 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
Message-ID: <CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>

Lets confirm the following statements now:

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use
G729 right now as it needs the license). By OUTGOING I mean call from
FreeSWITCH to your provider, and not back.

2. If #1 is true, then inbound-codec-negotiation is not much useful at this
point. Because this setting is for INCOMING leg, and not for OUTGOING. And
even if it _might_ be using GSM for INCOMING leg, but still transcode to
the OUTGOING PCMA/PCMU by some reason (this is what we need to find out -
why).

3. Try using absolute_codec_string before bridging your call to your SIP
provider:

<action application="set" data="absolute_codec_string=GSM"/>
<action application="bridge"
data="sofia/gateway/your_provider/12345678901_example"/>


And reply back if this changes anything...

Vitalie

2012/4/11 Malay Thakershi <mthakershi at gmail.com>

> Under my sipinterface_1 profile that I use for these calls, I have:
>
> <param name="inbound-codec-negotiation" value="greedy"/>
>
> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>
>> On Wed, 11 Apr 2012 13:31:45 -0500
>> Malay Thakershi <mthakershi at gmail.com> wrote:
>>
>> > Understood your point about G722. But if my list has GSM at higher
>> priority
>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call
>> using
>> > GSM?
>> >
>> > I don't know where G726 comparisons are coming from. If I haven't
>> mentioned
>> > it in vars.xml and far-end SDP doesn't support it then why would it even
>> > consider that codec?
>>
>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> (inbound-codec-negotiation):
>> 'generous' permits the remote codec list have precedence and 'win'
>>   the codec negotiation and selection process
>>
>> JY
>> --
>> Don't drop acid -- take it pass/fail.
>>                -- Seen in a Ladies' Room at Harvard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr 12 01:26:31 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 12 Apr 2012 00:26:31 +0300
Subject: [Freeswitch-users] Mach a pattern to route a group of dids.
In-Reply-To: <CADrcvV-0i4_nq682UpmfS+kGziTFchRPuL+G-=n7XUv3PbqcZA@mail.gmail.com>
References: <CADrcvV-0i4_nq682UpmfS+kGziTFchRPuL+G-=n7XUv3PbqcZA@mail.gmail.com>
Message-ID: <CALNZuwn=1TexFBcPh1uP_7gjZfTmFYe2uBJY3fpzOCdJ3Er2wg@mail.gmail.com>

By using a regex http://wiki.freeswitch.org/wiki/Regex for your condition
tags (e.g.
http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_1:_Matching_a_condition
)

-Avi



On Wed, Apr 11, 2012 at 11:53 PM, Cesar Bermudez <cesar.bermudez at gmail.com>
wrote:
> Hi FS'users
>
> I have a block of a did numbers.
>  example: 1111 0000 to 1111 2000
> i want to route from 1111 0000 to 1111 0099 to another server.
> and from 1111 0100 to 1111 2000 to another server.
> how can do that in the dialplan?
>
> best regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mthakershi at gmail.com  Thu Apr 12 01:47:23 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 11 Apr 2012 16:47:23 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
Message-ID: <CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>

> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use
> G729 right now as it needs the license). By OUTGOING I mean call from
> FreeSWITCH to your provider, and not back.
Yes. I want to try some codec that will use less bandwidth and see how
much audio quality is compromised. So I chose GSM because G729 has
license requirements.

I want this to work when someone calls my DID or my FS program calls
someone via DID. So both for incoming calls as well as outgoing calls.

> 2. If #1 is true, then?inbound-codec-negotiation is not much useful at
> this point. Because this setting is for INCOMING leg, and not for
> OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but still
> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to
> find out - why).
I am not very familiar with internals of FS so having lot of trouble
figuring this out.

Weird thing is, even if I do not have any mention of G729, it is being
negotiated and used. Here are my configuration lines:
vars.xml
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>

sip profile has:
<param name="inbound-codec-negotiation" value="greedy"/>

dial plan
  	<extension name="CH">
      <condition field="destination_number" expression="^105\d$">
		<action application="set" data="absolute_codec_string=GSM"/>
		<action application="sleep" data="1000"/>
		<action application="managed" data="clsAsmtApp"/>
      </condition>
	  </extension>

	
  <extension name="vitel-outbound">
   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
	 <action application="set" data="absolute_codec_string=GSM"/>
     <action application="set" data="effective_caller_id_number=18770001111"/>
     <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/>
    </condition>
  </extension>

> 3. Try using absolute_codec_string before bridging your call to your SIP
> provider:
>
> <action application="set" data="absolute_codec_string=GSM"/>
> <action application="bridge"
> data="sofia/gateway/your_provider/12345678901_example"/>
>
I did as you said. But found what I described above. If I remove
"greedy" setting, system simply uses PCMU codec like before.

This thing is so confusing. Even my SIP provider is saying they can't
help out with FS because some guy quit who knew the configurations.

Thanks for help.

On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
>
> Lets confirm the following statements now:
>
> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use
> G729 right now as it needs the license). By OUTGOING I mean call from
> FreeSWITCH to your provider, and not back.
>
> 2. If #1 is true, then?inbound-codec-negotiation is not much useful at
> this point. Because this setting is for INCOMING leg, and not for
> OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but still
> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to
> find out - why).
>
> 3. Try using absolute_codec_string before bridging your call to your SIP
> provider:
>
> <action application="set" data="absolute_codec_string=GSM"/>
> <action application="bridge"
> data="sofia/gateway/your_provider/12345678901_example"/>
>
>
> And reply back if this changes anything...
>
> Vitalie
>
> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>
>> Under my sipinterface_1 profile that I use for these calls, I have:
>>
>> <param name="inbound-codec-negotiation" value="greedy"/>
>>
>> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>>>
>>> On Wed, 11 Apr 2012 13:31:45 -0500
>>> Malay Thakershi <mthakershi at gmail.com> wrote:
>>>
>>> > Understood your point about G722. But if my list has GSM at higher
>>> > priority
>>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call
>>> > using
>>> > GSM?
>>> >
>>> > I don't know where G726 comparisons are coming from. If I haven't
>>> > mentioned
>>> > it in vars.xml and far-end SDP doesn't support it then why would it
>>> > even
>>> > consider that codec?
>>>
>>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>> (inbound-codec-negotiation):
>>> 'generous' permits the remote codec list have precedence and 'win'
>>> ? the codec negotiation and selection process
>>>
>>> JY
>>> --
>>> Don't drop acid -- take it pass/fail.
>>> ? ? ? ? ? ? ? ?-- Seen in a Ladies' Room at Harvard
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From me at nevian.org  Thu Apr 12 02:21:10 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 12 Apr 2012 02:21:10 +0400
Subject: [Freeswitch-users] custom bridge ringback
Message-ID: <36321334182870@web57.yandex.ru>

Hello,

I trying to sequental bridge w/custom ringback. No RB here - just silence till connect, definitely I did something wrong

bridge {ringback=/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav,effective_caller_id_number=${caller_id_number},ignore_early_media=true}[leg_timeout=30]loopback/7926xxxxxxx/allcoms.ru|loopback/7964yyyyyyy/allcoms.ru"

If I remove ignore_early_media - it plays media from leg B and not switching to next target as expected.
Loopback used as workaround for FS-3109 - can we use any other method?

btw if I use TGML in ringback, channel immediately HANGUP

2012-04-12 01:19:24.325555 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [^^:%(800:3200:425:0)]
2012-04-12 01:19:24.345530 [DEBUG] switch_core_codec.c:241 loopback/7964yyyyyyy-b Restore previous codec PCMA:8.
2012-04-12 01:19:24.345530 [DEBUG] switch_channel.c:2848 (sofia/external/7964yyyyyyy) Callstate Change EARLY -> HANGUP
2012-04-12 01:19:24.345530 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/external/7964yyyyyyy [CS_CONSUME_MEDIA] [NO_ANSWER]
2012-04-12 01:19:24.345530 [DEBUG] switch_channel.c:2871 Send signal sofia/external/7964yyyyyyy [KILL]

-- 
wbr,
Serge


From vetali100 at gmail.com  Thu Apr 12 02:48:45 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 11 Apr 2012 15:48:45 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
	<CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
Message-ID: <CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>

Lets forget for a minute about vars.xml and what is there (when you use
absolute_codec_string=GSM, it overrides any settings from vars.xml), and
concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.

Try to change your bridge to the following string:
<action application="bridge">
data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>

(obviously use your sip provider and phone number :)

And do one outgoing call to this sip provider.

Then, reply back whether the call was successful or not (did you hear the
voice and whether the other party did hear you). And paste the log to the
pastebin (http://pastebin.freeswitch.org/) and reply back with the link.





2012/4/11 Malay Thakershi <mthakershi at gmail.com>

> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use
> > G729 right now as it needs the license). By OUTGOING I mean call from
> > FreeSWITCH to your provider, and not back.
> Yes. I want to try some codec that will use less bandwidth and see how
> much audio quality is compromised. So I chose GSM because G729 has
> license requirements.
>
> I want this to work when someone calls my DID or my FS program calls
> someone via DID. So both for incoming calls as well as outgoing calls.
>
> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at
> > this point. Because this setting is for INCOMING leg, and not for
> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still
> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need
> to
> > find out - why).
> I am not very familiar with internals of FS so having lot of trouble
> figuring this out.
>
> Weird thing is, even if I do not have any mention of G729, it is being
> negotiated and used. Here are my configuration lines:
> vars.xml
>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>
> sip profile has:
> <param name="inbound-codec-negotiation" value="greedy"/>
>
> dial plan
>        <extension name="CH">
>      <condition field="destination_number" expression="^105\d$">
>                 <action application="set"
> data="absolute_codec_string=GSM"/>
>                 <action application="sleep" data="1000"/>
>                <action application="managed" data="clsAsmtApp"/>
>      </condition>
>          </extension>
>
>
>  <extension name="vitel-outbound">
>   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>          <action application="set" data="absolute_codec_string=GSM"/>
>      <action application="set"
> data="effective_caller_id_number=18770001111"/>
>     <action application="bridge"
> data="sofia/gateway/vitelity-outbound/$1"/>
>    </condition>
>  </extension>
>
> > 3. Try using absolute_codec_string before bridging your call to your SIP
> > provider:
> >
> > <action application="set" data="absolute_codec_string=GSM"/>
> > <action application="bridge"
> > data="sofia/gateway/your_provider/12345678901_example"/>
> >
> I did as you said. But found what I described above. If I remove
> "greedy" setting, system simply uses PCMU codec like before.
>
> This thing is so confusing. Even my SIP provider is saying they can't
> help out with FS because some guy quit who knew the configurations.
>
> Thanks for help.
>
> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
> wrote:
> >
> > Lets confirm the following statements now:
> >
> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use
> > G729 right now as it needs the license). By OUTGOING I mean call from
> > FreeSWITCH to your provider, and not back.
> >
> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at
> > this point. Because this setting is for INCOMING leg, and not for
> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still
> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need
> to
> > find out - why).
> >
> > 3. Try using absolute_codec_string before bridging your call to your SIP
> > provider:
> >
> > <action application="set" data="absolute_codec_string=GSM"/>
> > <action application="bridge"
> > data="sofia/gateway/your_provider/12345678901_example"/>
> >
> >
> > And reply back if this changes anything...
> >
> > Vitalie
> >
> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
> >>
> >> Under my sipinterface_1 profile that I use for these calls, I have:
> >>
> >> <param name="inbound-codec-negotiation" value="greedy"/>
> >>
> >> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
> >>>
> >>> On Wed, 11 Apr 2012 13:31:45 -0500
> >>> Malay Thakershi <mthakershi at gmail.com> wrote:
> >>>
> >>> > Understood your point about G722. But if my list has GSM at higher
> >>> > priority
> >>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call
> >>> > using
> >>> > GSM?
> >>> >
> >>> > I don't know where G726 comparisons are coming from. If I haven't
> >>> > mentioned
> >>> > it in vars.xml and far-end SDP doesn't support it then why would it
> >>> > even
> >>> > consider that codec?
> >>>
> >>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
> >>> (inbound-codec-negotiation):
> >>> 'generous' permits the remote codec list have precedence and 'win'
> >>>   the codec negotiation and selection process
> >>>
> >>> JY
> >>> --
> >>> Don't drop acid -- take it pass/fail.
> >>>                -- Seen in a Ladies' Room at Harvard
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Thu Apr 12 02:59:00 2012
From: mario_fs at mgtech.com (Mario G)
Date: Wed, 11 Apr 2012 15:59:00 -0700
Subject: [Freeswitch-users] custom bridge ringback
In-Reply-To: <36321334182870@web57.yandex.ru>
References: <36321334182870@web57.yandex.ru>
Message-ID: <57FFBE38-F6D3-4F54-B6DB-D35907DBB589@mgtech.com>

First, I am a FS novice here (17 months), but I do the same and I will tell you what I found:
1.  I use a separate statement prior to the bridge, I think yours would be (in fact you may just be missing the "file_string":
<action application="set" data="ringback=file_string:///usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav"/>
        
2. I recently found that mixing[] first then {} on the same bridge line caused problems, changing the first [] to {} so both sets matched fixed it. Yours is opposite though so I don't know if it applies.

May not be your issue but something to try.
Mario G

On Apr 11, 2012, at 3:21 PM, Serge S. Yuriev wrote:

> Hello,
> 
> I trying to sequental bridge w/custom ringback. No RB here - just silence till connect, definitely I did something wrong
> 
> bridge {ringback=/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav,effective_caller_id_number=${caller_id_number},ignore_early_media=true}[leg_timeout=30]loopback/7926xxxxxxx/allcoms.ru|loopback/7964yyyyyyy/allcoms.ru"
> 
> If I remove ignore_early_media - it plays media from leg B and not switching to next target as expected.
> Loopback used as workaround for FS-3109 - can we use any other method?
> 
> btw if I use TGML in ringback, channel immediately HANGUP
> 
> 2012-04-12 01:19:24.325555 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [^^:%(800:3200:425:0)]
> 2012-04-12 01:19:24.345530 [DEBUG] switch_core_codec.c:241 loopback/7964yyyyyyy-b Restore previous codec PCMA:8.
> 2012-04-12 01:19:24.345530 [DEBUG] switch_channel.c:2848 (sofia/external/7964yyyyyyy) Callstate Change EARLY -> HANGUP
> 2012-04-12 01:19:24.345530 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/external/7964yyyyyyy [CS_CONSUME_MEDIA] [NO_ANSWER]
> 2012-04-12 01:19:24.345530 [DEBUG] switch_channel.c:2871 Send signal sofia/external/7964yyyyyyy [KILL]
> 
> -- 
> wbr,
> Serge
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From kris at kriskinc.com  Thu Apr 12 05:24:28 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 11 Apr 2012 21:24:28 -0400
Subject: [Freeswitch-users] Pass the Regiseration message from SIP UA to
 remote SIP server
In-Reply-To: <CAJUJwtgvK6mrnw-mGs9fMipOxNN1sQrwiatK=kMU0AtkrEV94Q@mail.gmail.com>
References: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>
	<CAKV-jFyVsBRB3uKTTSME7FKiGnJrEvTj5AKAyy5uo+FLxcXACQ@mail.gmail.com>
	<CAJUJwtgvK6mrnw-mGs9fMipOxNN1sQrwiatK=kMU0AtkrEV94Q@mail.gmail.com>
Message-ID: <CAKDfjgfFJy_O=-X1semmXcoRHj=L36k79vTns9-uEFQfJJEjLg@mail.gmail.com>

Precisely.  The OP should put a SER-something in front of his
FreeSWITCH instances and use the path module.  FreeSWITCH supports
receiving REGISTER requests with path support but it doesn't support
sending them (which makes perfect sense given its intended function).

On Wed, Apr 11, 2012 at 3:57 PM, SamyGo <govoiper at gmail.com> wrote:
> Hello,
> What i've understood is that you want the REGISTER requests to be relayed to
> some other server via FreeSWITCH ! I'd say use OpenSIPS or Kamailio (SIP
> proxies) to do this kind of magic.
>
> Regards,
> Sammy.

-- 
Kristian Kielhofner


From mitch.capper at gmail.com  Thu Apr 12 05:28:18 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 11 Apr 2012 18:28:18 -0700
Subject: [Freeswitch-users] Using TLS on Freeswitch with CsipSimple on
	Android
In-Reply-To: <00b501cd180a$0bdcc610$23965230$@gmail.com>
References: <004401cd17df$c08a08f0$419e1ad0$@gmail.com>
	<CALn233oZ-WmwE8og76bx8G8kUWYQmpA1ZZSABvt3uMzRib6t5A@mail.gmail.com>
	<00b501cd180a$0bdcc610$23965230$@gmail.com>
Message-ID: <CALn233q0ngrdRWqAGYhTz98XS6UugiBzLTQpT1ghfvFXGzUQDg@mail.gmail.com>

you said "Currently, I can get two CSipSimple endpoints to register
and signal using TLS (sslv23) but can't get incoming calls on either
of them from either of them."

I was asking if you try to make outgoing calls on those (which should
use TLS/SRTP)  does that work?

~Mitch


From anton.jugatsu at gmail.com  Thu Apr 12 12:11:28 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 12 Apr 2012 12:11:28 +0400
Subject: [Freeswitch-users] [Default Vanilla Configuration]
	Conference.conf.xml and $${hold_music}
Message-ID: <CANBhyPBnW0dfku_zdSGNwAb1OjmZn6=o+BQ_aQVCMAYgGVYcPQ@mail.gmail.com>

Today while testing some default conference features ( thus using context
"default" and all vanilla configs)  i've discovered that:

 <extension name="wb_conferences">
      <condition field="destination_number" expression="^(31\d{2})$">
        <action application="answer"/>
        <action application="conference" data="$1-${domain_name}@wideband"/>
      </condition>
    </extension>

In the profile "wideband" at conference.conf.xml <param name="moh-sound"
contains value="$${hold_music}" but this global channel variable is
defiened at vars.xml and is local_stream://moh/8000

<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh/8000"/>

So when i entered into wideband or uwb or even cdquality (it doesn't matter
because in each profile there is param-name "moh-sound" contains
$${hold_music}) conference i'am will be listening moh/8000.

To listen moh/16000, moh/32000, moh/48000 i should manually adjust
corresponding music streaming quality, i.e for example for wb-conference
<param name="moh-sound" value="local_stream://moh/16000"/>

But there is another issue :) If do not install {hd,uhd,cd}-moh (assuming
that moh/8000 i do install) therefore my $${sounds_dir} will be containing
only 8000 stuff and when i will be enter one of hd/cd quality conferences i
would get this error

mod_local_stream.c:393 Unknown source moh/32000, trying 'default'

and thus listening default moh ($${sounds_dir}/music/8000).

What would be better: to get this error or manually additing each profile
at conference.conf.xml for listen hold-music @ HD-quality, there is the
question :)

I REPEAT I'm talking about default configuratin (i.e. vanilla) not
production cases :)
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From mytemike72 at gmail.com  Thu Apr 12 12:10:37 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 12 Apr 2012 10:10:37 +0200
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
Message-ID: <03c601cd1883$bffd66d0$3ff83470$@com>

Hi Guys,

 

I am in an urgent matter an need help with something, I am using an ESL
inbound connection to setup a new call.

It works fine on successful answered calls.

 

I am using the uuid_getvar api function whilst dialing to get the
endpoint_disposition of the dialing uuid to check when a call is answered.

So when it's answered it will return "ANSWER".

 

The problem with this is that if an attempt fails, the endpoint disposition
cannot be read with the uuid_getvar function because the dialing session is
already gone.

 

What is the best way to get the result in realtime?, I really need to know
the reason it failed.

 

 

Thanks for your help,

Mike.

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From peter.olsson at visionutveckling.se  Thu Apr 12 12:24:13 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 12 Apr 2012 08:24:13 +0000
Subject: [Freeswitch-users] [Default Vanilla
	Configuration]	Conference.conf.xml and $${hold_music}
Message-ID: <1FFF97C269757C458224B7C895F35F1508EC7E@cantor.std.visionutv.se>

Default in vars.xml is this - som you must have changed this. FS will append the correct sampling for you.

<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anton Kvashenkin
Skickat: den 12 april 2012 10:11
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] [Default Vanilla Configuration] Conference.conf.xml and $${hold_music}

Today while testing some default conference features ( thus using context "default" and all vanilla configs)  i've discovered that:

 <extension name="wb_conferences">
      <condition field="destination_number" expression="^(31\d{2})$">
        <action application="answer"/>
        <action application="conference" data="$1-${domain_name}@wideband"/>
      </condition>
    </extension>

In the profile "wideband" at conference.conf.xml <param name="moh-sound" contains value="$${hold_music}" but this global channel variable is defiened at vars.xml and is local_stream://moh/8000

<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh/8000"/>

So when i entered into wideband or uwb or even cdquality (it doesn't matter because in each profile there is param-name "moh-sound" contains $${hold_music}) conference i'am will be listening moh/8000.

To listen moh/16000, moh/32000, moh/48000 i should manually adjust corresponding music streaming quality, i.e for example for wb-conference
<param name="moh-sound" value="local_stream://moh/16000"/>

But there is another issue :) If do not install {hd,uhd,cd}-moh (assuming that moh/8000 i do install) therefore my $${sounds_dir} will be containing only 8000 stuff and when i will be enter one of hd/cd quality conferences i would get this error

mod_local_stream.c:393 Unknown source moh/32000, trying 'default'

and thus listening default moh ($${sounds_dir}/music/8000).

What would be better: to get this error or manually additing each profile at conference.conf.xml for listen hold-music @ HD-quality, there is the question :)

I REPEAT I'm talking about default configuratin (i.e. vanilla) not production cases :)


!DSPAM:4f868d9b32761309479703!
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From peter.olsson at visionutveckling.se  Thu Apr 12 12:25:41 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 12 Apr 2012 08:25:41 +0000
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
Message-ID: <1FFF97C269757C458224B7C895F35F1508EC89@cantor.std.visionutv.se>

Use the variable "origination_uuid", and set to an uuid that you know. Then it's possible to detect the events showing up for this channel uuid.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

Hi Guys,

I am in an urgent matter an need help with something, I am using an ESL inbound connection to setup a new call.
It works fine on successful answered calls.

I am using the uuid_getvar api function whilst dialing to get the endpoint_disposition of the dialing uuid to check when a call is answered.
So when it's answered it will return "ANSWER".

The problem with this is that if an attempt fails, the endpoint disposition cannot be read with the uuid_getvar function because the dialing session is already gone.

What is the best way to get the result in realtime?, I really need to know the reason it failed.


Thanks for your help,
Mike.
!DSPAM:4f868d9c32761957721393!
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From mytemike72 at gmail.com  Thu Apr 12 12:41:01 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 12 Apr 2012 10:41:01 +0200
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508EC89@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1508EC89@cantor.std.visionutv.se>
Message-ID: <03dd01cd1887$ff00da70$fd028f50$@com>

I am allready using the origination_uuid and pass it with the originate
command. I only create one now by using the create_uuid function.

 

So you say, don?t create a new one and just use one of a session that
already exists and is in progress? (so actually the session that is
?connected? to the inbound ESL session?  The originate/or system would not
have a problem with that? I would be setting up two calls wioth the same
UUID 
 and how would my other uuid functions respond to that? I can assume a
uuid kill would suddenly kill two legs?

 

Thanks,

Mike.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

Use the variable ?origination_uuid?, and set to an uuid that you know. Then
it?s possible to detect the events showing up for this channel uuid.

 

/Peter

 

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

 

Hi Guys,

 

I am in an urgent matter an need help with something, I am using an ESL
inbound connection to setup a new call.

It works fine on successful answered calls.

 

I am using the uuid_getvar api function whilst dialing to get the
endpoint_disposition of the dialing uuid to check when a call is answered.

So when it?s answered it will return ?ANSWER?.

 

The problem with this is that if an attempt fails, the endpoint disposition
cannot be read with the uuid_getvar function because the dialing session is
already gone.

 

What is the best way to get the result in realtime?, I really need to know
the reason it failed.

 

 

Thanks for your help,

Mike.

!DSPAM:4f868d9c32761957721393! 

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From me at nevian.org  Thu Apr 12 13:01:47 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 12 Apr 2012 13:01:47 +0400
Subject: [Freeswitch-users] custom bridge ringback
In-Reply-To: <57FFBE38-F6D3-4F54-B6DB-D35907DBB589@mgtech.com>
References: <36321334182870@web57.yandex.ru>
	<57FFBE38-F6D3-4F54-B6DB-D35907DBB589@mgtech.com>
Message-ID: <895321334221307@web69.yandex.ru>

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From dinhthuc1303 at gmail.com  Thu Apr 12 13:08:46 2012
From: dinhthuc1303 at gmail.com (dinh thuc)
Date: Thu, 12 Apr 2012 16:08:46 +0700
Subject: [Freeswitch-users] detect tone
Message-ID: <CAKMOEZncpPz62Kpj3tK2X29hpC8UYCt_atH2kEgP-8_8pHpcYQ@mail.gmail.com>

hi

I have try to build mod apllicatiion to dectect tone like mod_spandsp and I
have error below:

*2012-04-12 16:04:18.194204 [CRIT] switch_loadable_module.c:1300 Error
Loading module /usr/local/freeswitch/mod/mod_tone.so
**/usr/local/freeswitch/mod/mod_tone.so: undefined symbol:
super_tone_rx_release***


Makefile:

BASE=../../../..
LOCAL_OBJS=common.o
LOCAL_LDFLAGS = -ljpeg -lz -lutil
include $(BASE)/build/molo    dmake.rules

Please tell me these lib that I need to build application?


B&R

Dinh Thuc
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From anton.jugatsu at gmail.com  Thu Apr 12 13:20:56 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 12 Apr 2012 13:20:56 +0400
Subject: [Freeswitch-users] [Default Vanilla Configuration]
 Conference.conf.xml and $${hold_music}
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508EC7E@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1508EC7E@cantor.std.visionutv.se>
Message-ID: <CANBhyPBNf8OYbQVQdqrKz3-T7wacu2LPcFHUVzR3zxGVN6m-iA@mail.gmail.com>

Exactly. I screwed up, my apologies :)

12 ?????? 2012 ?. 12:24 ???????????? Peter Olsson <
peter.olsson at visionutveckling.se> ???????:

>  Default in vars.xml is this ? som you must have changed this. FS will
> append the correct sampling for you.****
>
> ** **
>
> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>****
>
> ** **
>
> /Peter****
>
> ** **
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anton Kvashenkin
> *Skickat:* den 12 april 2012 10:11
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] [Default Vanilla Configuration]
> Conference.conf.xml and $${hold_music}****
>
> ** **
>
> Today while testing some default conference features ( thus using context
> "default" and all vanilla configs)  i've discovered that:****
>
> ** **
>
>  <extension name="wb_conferences">****
>
>       <condition field="destination_number" expression="^(31\d{2})$">****
>
>         <action application="answer"/>****
>
>         <action application="conference" data="$1-${domain_name}@wideband
> "/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> In the profile "wideband" at conference.conf.xml <param name="moh-sound"
> contains value="$${hold_music}" but this global channel variable is
> defiened at vars.xml and is local_stream://moh/8000****
>
> ** **
>
> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh/8000"/>****
>
> ** **
>
> So when i entered into wideband or uwb or even cdquality (it doesn't
> matter because in each profile there is param-name "moh-sound" contains
> $${hold_music}) conference i'am will be listening moh/8000.****
>
> ** **
>
> To listen moh/16000, moh/32000, moh/48000 i should manually adjust
> corresponding music streaming quality, i.e for example for wb-conference**
> **
>
> <param name="moh-sound" value="local_stream://moh/16000"/>****
>
> ** **
>
> But there is another issue :) If do not install {hd,uhd,cd}-moh (assuming
> that moh/8000 i do install) therefore my $${sounds_dir} will be containing
> only 8000 stuff and when i will be enter one of hd/cd quality conferences i
> would get this error****
>
> ** **
>
> mod_local_stream.c:393 Unknown source moh/32000, trying 'default'****
>
> ** **
>
> and thus listening default moh ($${sounds_dir}/music/8000).****
>
> ** **
>
> What would be better: to get this error or manually additing each profile
> at conference.conf.xml for listen hold-music @ HD-quality, there is the
> question :)****
>
> ** **
>
> I REPEAT I'm talking about default configuratin (i.e. vanilla) not
> production cases :)****
>
> ** **
>
> ** **
>
> !DSPAM:4f868d9b32761309479703! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From peter.olsson at visionutveckling.se  Thu Apr 12 13:34:20 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 12 Apr 2012 09:34:20 +0000
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
Message-ID: <1FFF97C269757C458224B7C895F35F1508EE32@cantor.std.visionutv.se>

If you're already creating a uuid manually thats just fine (no, you can not use the same uuid for two channels). Now you just need to parse the events for that channel when you start the originate. As soon as the originate does any progress events will report this, using the uuid that you have defined.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:41
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

I am allready using the origination_uuid and pass it with the originate command. I only create one now by using the create_uuid function.

So you say, don't create a new one and just use one of a session that already exists and is in progress? (so actually the session that is 'connected' to the inbound ESL session?  The originate/or system would not have a problem with that? I would be setting up two calls wioth the same UUID ... and how would my other uuid functions respond to that? I can assume a uuid kill would suddenly kill two legs?

Thanks,
Mike.


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Peter Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

Use the variable "origination_uuid", and set to an uuid that you know. Then it's possible to detect the events showing up for this channel uuid.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

Hi Guys,

I am in an urgent matter an need help with something, I am using an ESL inbound connection to setup a new call.
It works fine on successful answered calls.

I am using the uuid_getvar api function whilst dialing to get the endpoint_disposition of the dialing uuid to check when a call is answered.
So when it's answered it will return "ANSWER".

The problem with this is that if an attempt fails, the endpoint disposition cannot be read with the uuid_getvar function because the dialing session is already gone.

What is the best way to get the result in realtime?, I really need to know the reason it failed.


Thanks for your help,
Mike.

!DSPAM:4f8694bf32762366476345!
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From anton.jugatsu at gmail.com  Thu Apr 12 13:48:42 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 12 Apr 2012 13:48:42 +0400
Subject: [Freeswitch-users] external sip profile - registered users get
 Originate Failed. Cause: USER_NOT_REGISTERED
In-Reply-To: <CAFArqsaUS_kQNNwL9LHRk9Ztq6vJ-b96W3qiHaQeGD==i3nGxA@mail.gmail.com>
References: <CAFArqsaUS_kQNNwL9LHRk9Ztq6vJ-b96W3qiHaQeGD==i3nGxA@mail.gmail.com>
Message-ID: <CANBhyPAi0Ciu3ybpTX9caPGk_5ysK9OXKO=Sqky6Wo=XXgzmWw@mail.gmail.com>

You are using the same port 5060 for internal and external profile, that's
not good :) Why did you change default 5080 port for external profile?

11 ?????? 2012 ?. 17:42 ???????????? Karsten Horsmann
<khorsmann at gmail.com>???????:

> Hi all,
>
>
> i try to install FS as SBC between public internet and private subnet.
> (SIP-User)--->[public-ip]FreeSWITCH[private-ip]--->(IVR System)
>
> If i try to reach an user, i got USER_NOT_REGISTERED. If i registered
> to the internal sip profile with the private address, everything works
> fine.
>
> Any hints?
>
>
> freeswitch at kamailio1> sofia status profile external
>
> =================================================================================================
> Name                    external
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_external
> Pres Hosts
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  212.xx.xx.xx
> SIP-IP                  212.xx.xx.xx
> URL                     sip:mod_sofia at 212.xx.xx.xx:5060
> BIND-URL                sip:mod_sofia at 212.xx.xx.xx:5060
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               PCMU,PCMA,GSM
> CODECS OUT              PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                4
> FAILED-CALLS-IN         2
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
> REGISTRATIONS           2
>
>
> freeswitch at kamailio1> sofia status profile internal
>
> =================================================================================================
> Name                    internal
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_internal
> Pres Hosts              212.59.42.189,172.20.100.74
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  172.20.100.74
> SIP-IP                  172.20.100.74
> URL                     sip:mod_sofia at 172.20.100.74:5060
> BIND-URL                sip:mod_sofia at 172.20.100.74:5060
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               PCMU,PCMA,GSM
> CODECS OUT              PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               1
> FAILED-CALLS-OUT        1
> REGISTRATIONS           0
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From khorsmann at gmail.com  Thu Apr 12 14:51:39 2012
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Thu, 12 Apr 2012 12:51:39 +0200
Subject: [Freeswitch-users] external sip profile - registered users get
 Originate Failed. Cause: USER_NOT_REGISTERED
In-Reply-To: <CANBhyPAi0Ciu3ybpTX9caPGk_5ysK9OXKO=Sqky6Wo=XXgzmWw@mail.gmail.com>
References: <CAFArqsaUS_kQNNwL9LHRk9Ztq6vJ-b96W3qiHaQeGD==i3nGxA@mail.gmail.com>
	<CANBhyPAi0Ciu3ybpTX9caPGk_5ysK9OXKO=Sqky6Wo=XXgzmWw@mail.gmail.com>
Message-ID: <CAFArqsby4u24L4RbFjLN-N-y1aq9QUVpQyNAEJU_T5kk=gs-EA@mail.gmail.com>

Hi Anton,

because all clients using 5060 as default and i didnt want to inform 1000 users
that they must use an diffrent port.

Also i use diffrent ip addresses for internal and external, so it
should be clear
where it comes from.

So what can i do to reconnice FS the registered user from external?


2012/4/12 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> You are using the same port 5060 for internal and external profile, that's
> not good :) Why did you change default 5080 port for external profile?


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*


From joohny at mail.ru  Thu Apr 12 10:49:30 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Thu, 12 Apr 2012 10:49:30 +0400
Subject: [Freeswitch-users] =?utf-8?q?voice_recordings_by_mod=5Fcallcenter?=
Message-ID: <E1SIDqV-0001pi-1K.joohny-mail-ru@f153.mail.ru>

Hello, colleagues.
I have some problems with voice recordings by mod_callcenter. If I route my call to bridge application the recording is good. If I route my call to callcenter application - recording is no good. I have distorted voice in one channel. It seems
that silence slices presents in recordings with some period. See attachment.?
FreeSwitch was downloaded from GIT month ago. It works on windows server 32bit 2003. Recordings is in wav format.
How can I solve this problem. Does someone have such things?
Thank you and sorry for bad English.
Evginey.
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From fdelawarde at wirelessmundi.com  Thu Apr 12 16:53:40 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 12 Apr 2012 14:53:40 +0200
Subject: [Freeswitch-users] voice recordings by mod_callcenter
In-Reply-To: <E1SIDqV-0001pi-1K.joohny-mail-ru@f153.mail.ru>
References: <E1SIDqV-0001pi-1K.joohny-mail-ru@f153.mail.ru>
Message-ID: <1334235220.20959.14.camel@luna.madrid.commsmundi.com>

Could your problem be related to this issue?

http://jira.freeswitch.org/browse/FS-4070

Fran?ois.


On Thu, 2012-04-12 at 10:49 +0400, ??????? wrote:
> Hello, colleagues.
> I have some problems with voice recordings by mod_callcenter. If I
> route my call to bridge application the recording is good. If I route
> my call to callcenter application - recording is no good. I have
> distorted voice in one channel. It seems
> that silence slices presents in recordings with some period. See
> attachment. 
> FreeSwitch was downloaded from GIT month ago. It works on windows
> server 32bit 2003. Recordings is in wav format.
> How can I solve this problem. Does someone have such things?
> 
> Thank you and sorry for bad English.
> 
> 
> Evginey. 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ocset at the800group.com  Thu Apr 12 17:59:31 2012
From: ocset at the800group.com (ocset)
Date: Thu, 12 Apr 2012 21:59:31 +0800
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <CAKnNFz-sF2xT5HgRROVG6eiYGUwqGaK_PK5r2q8FvyRDV9kiuw@mail.gmail.com>
References: <4F83C305.7070606@the800group.com>
	<CAKnNFz-sF2xT5HgRROVG6eiYGUwqGaK_PK5r2q8FvyRDV9kiuw@mail.gmail.com>
Message-ID: <4F86DFC3.4000109@the800group.com>

Hi Chris

I just took a chance and bought one from China - $50. This is not the 
exact model that I bought but it looks identical (mine was not branded 
with vodafone or sagem).

http://www.aliexpress.com/product-gs/477726489-GSM-Auto-Dialer-Fixed-GSM-Gateway-Fixed-Wireless-Terminal-Free-shipping-wholesalers.html

It seems to work fine but does not show signal strength so I am going to 
try one that has an indicator next.

http://www.aliexpress.com/product-gs/396284880-5-years-manufacturer-1-year-warranty-GSM-gateway-GSM-FCT-fixed-cellular-terminal-GSM-FWT-fixed-wholesalers.html

Regards

On 04/12/2012 05:01 AM, chris wrote:
> what gsm gateway are you using? ive been looking for a good one
>
> On Tue, Apr 10, 2012 at 1:20 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
>     The first line is a POTS line and the second line is a GSM gateway
>     device. Both devices work correctly if I dial out but when I try and
>     bridge between the two, nothing works as expected. The bridge is from
>     the POTS line to a mobile number using the GSM gateway.
>
>     I have created the following bridge for the POTS line (phone numbers
>     removed for privacy reasons)
>
>     <extension name="pots_line">
>     <condition field="destination_number" expression="^(xxxxxxxxxx)$">
>     <action application="bridge"
>     data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062
>     <http://zzzzzzzzzz at 192.168.0.160:5062>"/>
>     </condition>
>     </extension>
>
>     The bridge does eventually happen (long delay) but neither party can
>     hear each other. Also, when I hang-up with one or both phones, the
>     connection does not terminate and I have to reboot the GXW4104 to
>     break
>     the connection.
>
>     I have bought the Freeswitch books and this seems to be something that
>     should easily work. I have not added any variables into the
>     <extension>
>     definition as this is a simple bridge.
>
>     What have I missed?
>
>     Thanks in advance
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mytemike72 at gmail.com  Thu Apr 12 18:24:45 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 12 Apr 2012 16:24:45 +0200
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508EE32@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1508EE32@cantor.std.visionutv.se>
Message-ID: <041001cd18b8$035d07d0$0a171770$@com>

Peter, I know, I am doing this allready, that is how I check the disposition
while dialing already, I am using uuid_getvar to reach it.

The problem is, that when it is not answered, ie Busy or congestion, the
session is killed by the core and no longer exists.

So when it stops, I cannot do a uuid_getvar anymore as the session no longer
exists.

 

string cDialString = "{origination_uuid=" + thisDial.CallId_LegB // This my
generated uuid

+ ",origination_caller_id_number=" + thisDial.CLIP

+ ",origination_caller_id_name=" + thisDial.CLIP

+ "}sofia/external/" + thisExtension.Destination+'@'+ thisDial.Gateway

+ " &park()";

 

var eslEvent = thisFSConnection.Connection.Api("originate", cDialString);

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

 

Stopwatch dialTimeout = new Stopwatch();

dialTimeout.Start();

while (CallIdExists(thisFSConnection, thisCall.callId) &&
CallIdExists(thisFSConnection, thisDial.CallId_LegB) && dispo == "EARLY
MEDIA" && dialTimeout.ElapsedMilliseconds <
thisDial.DialAttemptDuration*1000 )

{

if (dispo != dispoLast)

{

dispoLast = dispo;

}

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

Thread.Sleep(100);

}

dialTimeout.Stop();

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

 

if (dispo == "ANSWER")

 

The last dispo is never filled correctly, because the session no longer
exist, unless it was answered


 

Thanks,

Mike.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 11:34
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

If you?re already creating a uuid manually thats just fine (no, you can not
use the same uuid for two channels). Now you just need to parse the events
for that channel when you start the originate. As soon as the originate does
any progress events will report this, using the uuid that you have defined.

 

/Peter

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:41
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

I am allready using the origination_uuid and pass it with the originate
command. I only create one now by using the create_uuid function.

 

So you say, don?t create a new one and just use one of a session that
already exists and is in progress? (so actually the session that is
?connected? to the inbound ESL session?  The originate/or system would not
have a problem with that? I would be setting up two calls wioth the same
UUID 
 and how would my other uuid functions respond to that? I can assume a
uuid kill would suddenly kill two legs?

 

Thanks,

Mike.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

Use the variable ?origination_uuid?, and set to an uuid that you know. Then
it?s possible to detect the events showing up for this channel uuid.

 

/Peter

 

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

 

Hi Guys,

 

I am in an urgent matter an need help with something, I am using an ESL
inbound connection to setup a new call.

It works fine on successful answered calls.

 

I am using the uuid_getvar api function whilst dialing to get the
endpoint_disposition of the dialing uuid to check when a call is answered.

So when it?s answered it will return ?ANSWER?.

 

The problem with this is that if an attempt fails, the endpoint disposition
cannot be read with the uuid_getvar function because the dialing session is
already gone.

 

What is the best way to get the result in realtime?, I really need to know
the reason it failed.

 

 

Thanks for your help,

Mike.

 

!DSPAM:4f8694bf32762366476345! 

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From lloyd.aloysius at gmail.com  Thu Apr 12 18:30:13 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Thu, 12 Apr 2012 10:30:13 -0400
Subject: [Freeswitch-users] Efficient way to query freeswitch - status and
	info
Message-ID: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>

Hi All,

I would like to find a better[efficient] way to query freeswitch. I
am implementing a dashboard [PHP/Jquery] to view status of freeswitch.Lets
assume system is a busy and always 100+ active calls.

I want to know the following info

1. show calls
2. Sofia status
3. show channels
etc

What is the better direction to implement this

a. Query Data Base tables

or

b. Event Library

or

?

Thanks in advanced.
Lloyd
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From wesleyakio at tuntscorp.com  Thu Apr 12 18:41:29 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Thu, 12 Apr 2012 11:41:29 -0300
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
In-Reply-To: <041001cd18b8$035d07d0$0a171770$@com>
References: <1FFF97C269757C458224B7C895F35F1508EE32@cantor.std.visionutv.se>
	<041001cd18b8$035d07d0$0a171770$@com>
Message-ID: <CAGTsDSvpn2=fQa_Cm5hGLNEqMWJiJd3SWFH3tHU7E-6nA7hrng@mail.gmail.com>

I guess what peter means is
http://wiki.freeswitch.org/wiki/Event_Socket#event, but I might be wrong.

Best,

Wesley Akio
TuntsCorp.com


On Thu, Apr 12, 2012 at 11:24 AM, Michael Lutz <mytemike72 at gmail.com> wrote:

> Peter, I know, I am doing this allready, that is how I check the
> disposition while dialing already, I am using uuid_getvar to reach it.****
>
> The problem is, that when it is not answered, ie Busy or congestion, the
> session is killed by the core and no longer exists.****
>
> So when it stops, I cannot do a uuid_getvar anymore as the session no
> longer exists.****
>
> ** **
>
> string cDialString = "{origination_uuid=" + thisDial.CallId_LegB // This
> my generated uuid****
>
> + ",origination_caller_id_number=" + thisDial.CLIP****
>
> + ",origination_caller_id_name=" + thisDial.CLIP****
>
> + "}sofia/external/" + thisExtension.Destination+'@'+ thisDial.Gateway****
>
> + " &park()";****
>
> ** **
>
> var eslEvent = thisFSConnection.Connection.Api("originate", cDialString);*
> ***
>
> dispo = thisFSConnection.Connection.Api("uuid_getvar",
> thisDial.CallId_LegB + " endpoint_disposition").GetBody();****
>
> ** **
>
> Stopwatch dialTimeout = new Stopwatch();****
>
> dialTimeout.Start();****
>
> while (CallIdExists(thisFSConnection, thisCall.callId) &&
> CallIdExists(thisFSConnection, thisDial.CallId_LegB) && dispo == "EARLY
> MEDIA" && dialTimeout.ElapsedMilliseconds <
> thisDial.DialAttemptDuration*1000 )****
>
> {****
>
> if (dispo != dispoLast)****
>
> {****
>
> dispoLast = dispo;****
>
> }****
>
> dispo = thisFSConnection.Connection.Api("uuid_getvar",
> thisDial.CallId_LegB + " endpoint_disposition").GetBody();****
>
> Thread.Sleep(100);****
>
> }****
>
> dialTimeout.Stop();****
>
> dispo = thisFSConnection.Connection.Api("uuid_getvar",
> thisDial.CallId_LegB + " endpoint_disposition").GetBody();****
>
> ** **
>
> if (dispo == "ANSWER")****
>
> ** **
>
> The last dispo is never filled correctly, because the session no longer
> exist, unless it was answered?****
>
> ** **
>
> Thanks,****
>
> Mike.****
>
> ** **
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter Olsson
> *Sent:* donderdag 12 april 2012 11:34
>
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] ESL originate retreiving dialresult.****
>
> ** **
>
> If you?re already creating a uuid manually thats just fine (no, you can
> not use the same uuid for two channels). Now you just need to parse the
> events for that channel when you start the originate. As soon as the
> originate does any progress events will report this, using the uuid that
> you have defined.****
>
>  ****
>
> /Peter****
>
>  ****
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Michael Lutz
> *Skickat:* den 12 april 2012 10:41
> *Till:* 'FreeSWITCH Users Help'
> *?mne:* Re: [Freeswitch-users] ESL originate retreiving dialresult.****
>
>  ****
>
> I am allready using the origination_uuid and pass it with the originate
> command. I only create one now by using the create_uuid function.****
>
>  ****
>
> So you say, don?t create a new one and just use one of a session that
> already exists and is in progress? (so actually the session that is
> ?connected? to the inbound ESL session?  The originate/or system would not
> have a problem with that? I would be setting up two calls wioth the same
> UUID ? and how would my other uuid functions respond to that? I can assume
> a uuid kill would suddenly kill two legs?****
>
>  ****
>
> Thanks,****
>
> Mike.****
>
>  ****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Peter
> Olsson
> *Sent:* donderdag 12 april 2012 10:26
> *To:* 'FreeSWITCH Users Help'
> *Subject:* Re: [Freeswitch-users] ESL originate retreiving dialresult.****
>
>  ****
>
> Use the variable ?origination_uuid?, and set to an uuid that you know.
> Then it?s possible to detect the events showing up for this channel uuid.*
> ***
>
>  ****
>
> /Peter****
>
>  ****
>
>  ****
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Michael Lutz
> *Skickat:* den 12 april 2012 10:11
> *Till:* 'FreeSWITCH Users Help'
> *?mne:* [Freeswitch-users] ESL originate retreiving dialresult.****
>
>  ****
>
> Hi Guys,****
>
>  ****
>
> I am in an urgent matter an need help with something, I am using an ESL
> inbound connection to setup a new call.****
>
> It works fine on successful answered calls.****
>
>  ****
>
> I am using the uuid_getvar api function whilst dialing to get the
> endpoint_disposition of the dialing uuid to check when a call is answered.
> ****
>
> So when it?s answered it will return ?ANSWER?.****
>
>  ****
>
> The problem with this is that if an attempt fails, the endpoint
> disposition cannot be read with the uuid_getvar function because the
> dialing session is already gone.****
>
>  ****
>
> What is the best way to get the result in realtime?, I really need to know
> the reason it failed.****
>
>  ****
>
>  ****
>
> Thanks for your help,****
>
> Mike.****
>
>  ****
>
> !DSPAM:4f8694bf32762366476345! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From peter.olsson at visionutveckling.se  Thu Apr 12 18:49:56 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 12 Apr 2012 14:49:56 +0000
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
Message-ID: <1FFF97C269757C458224B7C895F35F1508F0F7@cantor.std.visionutv.se>

What you should do (and what my intention was to describe) is to change how you detect state changes to the call. Now you use a poll mechanism. If you changed this to listen to events instead, you would get the changes as soon as it happens, and you will be able to read all the variables from these events.

Also, "originate" will not return until media is present, so if you dial somewhere without early media, your existing code will hang until the call has been answered, to get around this you can use bpapi instead of Api, which will fire up a new thread in FS during the setup of the call and return immediately.

A short example;

1.       Send bgapi originate ..... to FS - this will return immediately.

2.       Start polling for events, as soon as the state of your new call changes an event will be sent, with the UUID that you specified.

3.       Read the variables in the event to get further information. Since the variables are contained in the event object, it won't matter if the call was disconnected already.

It will require basic changes in your code, but ESL was made to work this way (async with events, not semi-async with polling), and it will make things easier for you in the future.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 16:25
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

Peter, I know, I am doing this allready, that is how I check the disposition while dialing already, I am using uuid_getvar to reach it.
The problem is, that when it is not answered, ie Busy or congestion, the session is killed by the core and no longer exists.
So when it stops, I cannot do a uuid_getvar anymore as the session no longer exists.

string cDialString = "{origination_uuid=" + thisDial.CallId_LegB // This my generated uuid
+ ",origination_caller_id_number=" + thisDial.CLIP
+ ",origination_caller_id_name=" + thisDial.CLIP
+ "}sofia/external/" + thisExtension.Destination+'@'+ thisDial.Gateway
+ " &park()";

var eslEvent = thisFSConnection.Connection.Api("originate", cDialString);
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();

Stopwatch dialTimeout = new Stopwatch();
dialTimeout.Start();
while (CallIdExists(thisFSConnection, thisCall.callId) && CallIdExists(thisFSConnection, thisDial.CallId_LegB) && dispo == "EARLY MEDIA" && dialTimeout.ElapsedMilliseconds < thisDial.DialAttemptDuration*1000 )
{
if (dispo != dispoLast)
{
dispoLast = dispo;
}
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();
Thread.Sleep(100);
}
dialTimeout.Stop();
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();

if (dispo == "ANSWER")

The last dispo is never filled correctly, because the session no longer exist, unless it was answered...

Thanks,
Mike.


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Peter Olsson
Sent: donderdag 12 april 2012 11:34
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

If you're already creating a uuid manually thats just fine (no, you can not use the same uuid for two channels). Now you just need to parse the events for that channel when you start the originate. As soon as the originate does any progress events will report this, using the uuid that you have defined.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Michael Lutz
Skickat: den 12 april 2012 10:41
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

I am allready using the origination_uuid and pass it with the originate command. I only create one now by using the create_uuid function.

So you say, don't create a new one and just use one of a session that already exists and is in progress? (so actually the session that is 'connected' to the inbound ESL session?  The originate/or system would not have a problem with that? I would be setting up two calls wioth the same UUID ... and how would my other uuid functions respond to that? I can assume a uuid kill would suddenly kill two legs?

Thanks,
Mike.


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Peter Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

Use the variable "origination_uuid", and set to an uuid that you know. Then it's possible to detect the events showing up for this channel uuid.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

Hi Guys,

I am in an urgent matter an need help with something, I am using an ESL inbound connection to setup a new call.
It works fine on successful answered calls.

I am using the uuid_getvar api function whilst dialing to get the endpoint_disposition of the dialing uuid to check when a call is answered.
So when it's answered it will return "ANSWER".

The problem with this is that if an attempt fails, the endpoint disposition cannot be read with the uuid_getvar function because the dialing session is already gone.

What is the best way to get the result in realtime?, I really need to know the reason it failed.


Thanks for your help,
Mike.


!DSPAM:4f86e5a732761122815096!
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From peter.olsson at visionutveckling.se  Thu Apr 12 18:51:34 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 12 Apr 2012 14:51:34 +0000
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
Message-ID: <1FFF97C269757C458224B7C895F35F1508F105@cantor.std.visionutv.se>

Yes, that's what I meant :)

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Wesley Akio
Skickat: den 12 april 2012 16:41
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

I guess what peter means is  http://wiki.freeswitch.org/wiki/Event_Socket#event, but I might be wrong.

Best,

Wesley Akio
TuntsCorp.com

On Thu, Apr 12, 2012 at 11:24 AM, Michael Lutz <mytemike72 at gmail.com<mailto:mytemike72 at gmail.com>> wrote:
Peter, I know, I am doing this allready, that is how I check the disposition while dialing already, I am using uuid_getvar to reach it.
The problem is, that when it is not answered, ie Busy or congestion, the session is killed by the core and no longer exists.
So when it stops, I cannot do a uuid_getvar anymore as the session no longer exists.

string cDialString = "{origination_uuid=" + thisDial.CallId_LegB // This my generated uuid
+ ",origination_caller_id_number=" + thisDial.CLIP
+ ",origination_caller_id_name=" + thisDial.CLIP
+ "}sofia/external/" + thisExtension.Destination+'@'+ thisDial.Gateway
+ " &park()";

var eslEvent = thisFSConnection.Connection.Api("originate", cDialString);
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();

Stopwatch dialTimeout = new Stopwatch();
dialTimeout.Start();
while (CallIdExists(thisFSConnection, thisCall.callId) && CallIdExists(thisFSConnection, thisDial.CallId_LegB) && dispo == "EARLY MEDIA" && dialTimeout.ElapsedMilliseconds < thisDial.DialAttemptDuration*1000 )
{
if (dispo != dispoLast)
{
dispoLast = dispo;
}
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();
Thread.Sleep(100);
}
dialTimeout.Stop();
dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB + " endpoint_disposition").GetBody();

if (dispo == "ANSWER")

The last dispo is never filled correctly, because the session no longer exist, unless it was answered...

Thanks,
Mike.


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Peter Olsson
Sent: donderdag 12 april 2012 11:34

To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

If you're already creating a uuid manually thats just fine (no, you can not use the same uuid for two channels). Now you just need to parse the events for that channel when you start the originate. As soon as the originate does any progress events will report this, using the uuid that you have defined.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Michael Lutz
Skickat: den 12 april 2012 10:41
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

I am allready using the origination_uuid and pass it with the originate command. I only create one now by using the create_uuid function.

So you say, don't create a new one and just use one of a session that already exists and is in progress? (so actually the session that is 'connected' to the inbound ESL session?  The originate/or system would not have a problem with that? I would be setting up two calls wioth the same UUID ... and how would my other uuid functions respond to that? I can assume a uuid kill would suddenly kill two legs?

Thanks,
Mike.


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Peter Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

Use the variable "origination_uuid", and set to an uuid that you know. Then it's possible to detect the events showing up for this channel uuid.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

Hi Guys,

I am in an urgent matter an need help with something, I am using an ESL inbound connection to setup a new call.
It works fine on successful answered calls.

I am using the uuid_getvar api function whilst dialing to get the endpoint_disposition of the dialing uuid to check when a call is answered.
So when it's answered it will return "ANSWER".

The problem with this is that if an attempt fails, the endpoint disposition cannot be read with the uuid_getvar function because the dialing session is already gone.

What is the best way to get the result in realtime?, I really need to know the reason it failed.


Thanks for your help,
Mike.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f86e85032762051311084!
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From mitch.capper at gmail.com  Thu Apr 12 18:59:45 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 12 Apr 2012 07:59:45 -0700
Subject: [Freeswitch-users] Efficient way to query freeswitch - status
	and info
In-Reply-To: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
References: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
Message-ID: <CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>

Well using ESL to connect to FS you can then query the hash DB which
stores some of those items or use the xml status commands would
probably give you the best results.


~Mitch

On Thu, Apr 12, 2012 at 7:30 AM, Lloyd Aloysius
<lloyd.aloysius at gmail.com> wrote:
> Hi All,
>
> I would like to find a better[efficient] way to query freeswitch. I
> am?implementing?a?dashboard?[PHP/Jquery] to view status of freeswitch.Lets
> assume system is a busy and always 100+ active calls.
>
> I want to know the following info
>
> 1. show calls
> 2. Sofia status
> 3. show?channels
> etc
>
> What is the better direction to implement this
>
> a. Query Data Base tables
>
> or
>
> b. Event Library
>
> or
>
> ?
>
> Thanks in advanced.
> Lloyd
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lloyd.aloysius at gmail.com  Thu Apr 12 19:12:49 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Thu, 12 Apr 2012 11:12:49 -0400
Subject: [Freeswitch-users] Efficient way to query freeswitch - status
	and info
In-Reply-To: <CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
References: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
	<CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
Message-ID: <CAGvuWYZ+KFJadOMPfVPhvjJSg1P+jBO5St6AcBH0k0zt0MyC=w@mail.gmail.com>

Our core and all other information in mysql database. You mean hashdb is
Sqlite?

xmlstatus -- how to access from externally?

I find sometimes the registration tables have records . but the devices are
not connected to freeswitch?

What is reliable accurate way to get information in real time?

Lloyd



*  <http://www.sunteltech.ca/blog/>*



On Thu, Apr 12, 2012 at 10:59 AM, Mitch Capper <mitch.capper at gmail.com>wrote:

> Well using ESL to connect to FS you can then query the hash DB which
> stores some of those items or use the xml status commands would
> probably give you the best results.
>
>
> ~Mitch
>
> On Thu, Apr 12, 2012 at 7:30 AM, Lloyd Aloysius
> <lloyd.aloysius at gmail.com> wrote:
> > Hi All,
> >
> > I would like to find a better[efficient] way to query freeswitch. I
> > am implementing a dashboard [PHP/Jquery] to view status of
> freeswitch.Lets
> > assume system is a busy and always 100+ active calls.
> >
> > I want to know the following info
> >
> > 1. show calls
> > 2. Sofia status
> > 3. show channels
> > etc
> >
> > What is the better direction to implement this
> >
> > a. Query Data Base tables
> >
> > or
> >
> > b. Event Library
> >
> > or
> >
> > ?
> >
> > Thanks in advanced.
> > Lloyd
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr 12 19:14:32 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 12 Apr 2012 18:14:32 +0300
Subject: [Freeswitch-users] Efficient way to query freeswitch - status
	and info
In-Reply-To: <CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
References: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
	<CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
Message-ID: <CALNZuwnsfJ5R_7QerAo3WOvf+UqKkRdTWioBopwfCtm8NZMjMQ@mail.gmail.com>

If you subscribe on ESL to the various events for channel
start/transfer/end you'll have running updates on the calls, rather than
polling every X seconds.

-Avi
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From mytemike72 at gmail.com  Thu Apr 12 19:23:46 2012
From: mytemike72 at gmail.com (Michael Lutz)
Date: Thu, 12 Apr 2012 17:23:46 +0200
Subject: [Freeswitch-users] ESL originate retreiving dialresult.
In-Reply-To: <1FFF97C269757C458224B7C895F35F1508F0F7@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1508F0F7@cantor.std.visionutv.se>
Message-ID: <043301cd18c0$41bbb320$c5331960$@com>

And excellent explanation, and makes sense! :-)

Thanks Peter for your detailed explanation, I will change my code as you
suggested!

 

Regards,

Mike.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 16:50
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

What you should do (and what my intention was to describe) is to change how
you detect state changes to the call. Now you use a poll mechanism. If you
changed this to listen to events instead, you would get the changes as soon
as it happens, and you will be able to read all the variables from these
events.

 

Also, ?originate? will not return until media is present, so if you dial
somewhere without early media, your existing code will hang until the call
has been answered, to get around this you can use bpapi instead of Api,
which will fire up a new thread in FS during the setup of the call and
return immediately.

 

A short example;

1.       Send bgapi originate ..... to FS ? this will return immediately.

2.       Start polling for events, as soon as the state of your new call
changes an event will be sent, with the UUID that you specified.

3.       Read the variables in the event to get further information. Since
the variables are contained in the event object, it won?t matter if the call
was disconnected already.

 

It will require basic changes in your code, but ESL was made to work this
way (async with events, not semi-async with polling), and it will make
things easier for you in the future.

 

/Peter

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 16:25
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

Peter, I know, I am doing this allready, that is how I check the disposition
while dialing already, I am using uuid_getvar to reach it.

The problem is, that when it is not answered, ie Busy or congestion, the
session is killed by the core and no longer exists.

So when it stops, I cannot do a uuid_getvar anymore as the session no longer
exists.

 

string cDialString = "{origination_uuid=" + thisDial.CallId_LegB // This my
generated uuid

+ ",origination_caller_id_number=" + thisDial.CLIP

+ ",origination_caller_id_name=" + thisDial.CLIP

+ "}sofia/external/" + thisExtension.Destination+'@'+ thisDial.Gateway

+ " &park()";

 

var eslEvent = thisFSConnection.Connection.Api("originate", cDialString);

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

 

Stopwatch dialTimeout = new Stopwatch();

dialTimeout.Start();

while (CallIdExists(thisFSConnection, thisCall.callId) &&
CallIdExists(thisFSConnection, thisDial.CallId_LegB) && dispo == "EARLY
MEDIA" && dialTimeout.ElapsedMilliseconds <
thisDial.DialAttemptDuration*1000 )

{

if (dispo != dispoLast)

{

dispoLast = dispo;

}

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

Thread.Sleep(100);

}

dialTimeout.Stop();

dispo = thisFSConnection.Connection.Api("uuid_getvar", thisDial.CallId_LegB
+ " endpoint_disposition").GetBody();

 

if (dispo == "ANSWER")

 

The last dispo is never filled correctly, because the session no longer
exist, unless it was answered


 

Thanks,

Mike.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 11:34
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

If you?re already creating a uuid manually thats just fine (no, you can not
use the same uuid for two channels). Now you just need to parse the events
for that channel when you start the originate. As soon as the originate does
any progress events will report this, using the uuid that you have defined.

 

/Peter

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:41
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

I am allready using the origination_uuid and pass it with the originate
command. I only create one now by using the create_uuid function.

 

So you say, don?t create a new one and just use one of a session that
already exists and is in progress? (so actually the session that is
?connected? to the inbound ESL session?  The originate/or system would not
have a problem with that? I would be setting up two calls wioth the same
UUID 
 and how would my other uuid functions respond to that? I can assume a
uuid kill would suddenly kill two legs?

 

Thanks,

Mike.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter
Olsson
Sent: donderdag 12 april 2012 10:26
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] ESL originate retreiving dialresult.

 

Use the variable ?origination_uuid?, and set to an uuid that you know. Then
it?s possible to detect the events showing up for this channel uuid.

 

/Peter

 

 

Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Lutz
Skickat: den 12 april 2012 10:11
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] ESL originate retreiving dialresult.

 

Hi Guys,

 

I am in an urgent matter an need help with something, I am using an ESL
inbound connection to setup a new call.

It works fine on successful answered calls.

 

I am using the uuid_getvar api function whilst dialing to get the
endpoint_disposition of the dialing uuid to check when a call is answered.

So when it?s answered it will return ?ANSWER?.

 

The problem with this is that if an attempt fails, the endpoint disposition
cannot be read with the uuid_getvar function because the dialing session is
already gone.

 

What is the best way to get the result in realtime?, I really need to know
the reason it failed.

 

 

Thanks for your help,

Mike.

 

 

!DSPAM:4f86e5a732761122815096! 

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From mitch.capper at gmail.com  Thu Apr 12 19:32:27 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Thu, 12 Apr 2012 08:32:27 -0700
Subject: [Freeswitch-users] Efficient way to query freeswitch - status
	and info
In-Reply-To: <CALNZuwnsfJ5R_7QerAo3WOvf+UqKkRdTWioBopwfCtm8NZMjMQ@mail.gmail.com>
References: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
	<CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
	<CALNZuwnsfJ5R_7QerAo3WOvf+UqKkRdTWioBopwfCtm8NZMjMQ@mail.gmail.com>
Message-ID: <CALn233roTyAEHO_fCmx4XTjCaqVOgrTsDiO=Bsbbk9KnQf3fRw@mail.gmail.com>

Many of the status requests now there is an XML option also like
"sofia xmlstatus" you can use ESL to do that (see wiki for other
commands).  Using the events however is better for performance  to
avoid polling but does obviously require more logic to maintain state.

~Mitch


From joohny at mail.ru  Thu Apr 12 19:49:27 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Thu, 12 Apr 2012 19:49:27 +0400
Subject: [Freeswitch-users]
 =?utf-8?q?voice_recordings_by_mod=5Fcallcenter?=
In-Reply-To: <mailman.7093.1334235148.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.7093.1334235148.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <E1SIMH1-0007sQ-2Q.joohny-mail-ru@f210.mail.ru>

Hi
Thank you for url, but it is not the same. Quite different.
There is inserts of silence into one of channels. Same phones dialed by "bridge" has exelent recordings. If I dial by "callcenter" - one channel has problems. :( 
I can attach WAV to mail, or put url to download it. If somebody want to hear IT :)


Evginey.

> Could your problem be related to this issue?
> 
> http://jira.freeswitch.org/browse/FS-4070
> 
> Fran?ois.
> 
> 
> On Thu, 2012-04-12 at 10:49 +0400, ??????? wrote:
> > Hello, colleagues.
> > I have some problems with voice recordings by mod_callcenter. If I
> > route my call to bridge application the recording is good. If I route
> > my call to callcenter application - recording is no good. I have
> > distorted voice in one channel. It seems
> > that silence slices presents in recordings with some period. See
> > attachment. 
> > FreeSwitch was downloaded from GIT month ago. It works on windows
> > server 32bit 2003. Recordings is in wav format.
> > How can I solve this problem. Does someone have such things?
> > 
> > Thank you and sorry for bad English.
> > 
> > 
> > Evginey. 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org

From me at nevian.org  Thu Apr 12 22:22:20 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 12 Apr 2012 22:22:20 +0400
Subject: [Freeswitch-users] load mod_lcr error
In-Reply-To: <CAGKPtBu_sS_Z3FS9DzLmMUhAk2zJqm9-Xw_=uKE-nTYymukXpg@mail.gmail.com>
References: <CAGKPtBu_sS_Z3FS9DzLmMUhAk2zJqm9-Xw_=uKE-nTYymukXpg@mail.gmail.com>
Message-ID: <54691334254940@web108.yandex.ru>

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From anthony.minessale at gmail.com  Fri Apr 13 00:29:50 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Apr 2012 15:29:50 -0500
Subject: [Freeswitch-users] incorrect reg rows in sqlite database
	sofia_reg_internal.db
In-Reply-To: <SNT127-W64A638FB74DE9049E307D8D5350@phx.gbl>
References: <SNT127-W39E7937E807E9A7A0FEA1D5320@phx.gbl>
	<SNT127-W406A97A21171F08C78C39FD5320@phx.gbl>
	<SNT127-W65713C002F6E87F9869B13D5320@phx.gbl>
	<CAKbxfG8bYVwe+LFkDoa6tWA2KoF3V8vy4b8iE6m6mywWvdGDPw@mail.gmail.com>
	<SNT127-W64A638FB74DE9049E307D8D5350@phx.gbl>
Message-ID: <CAKbxfG-gxmjT2iaRz=0A3yooHwrb1Qsji=ywWOAxYBzZCL2oLw@mail.gmail.com>

also you can just delete it and it will auto-generate a new one.


On Wed, Apr 11, 2012 at 7:05 AM, Yuriy Nasida <nasida at live.ru> wrote:
> Thank again.
>
> I have fixed this issue.
> I just have replaced sqlite db file sofia_reg_internal.db by same file from
> other FS and have done flush all regs.
> (usual flushing didn't help)
>
> It Looks like this db file was damaged not sure how exactly but replacement
> fixed this issue.
>
>
>> Date: Wed, 4 Apr 2012 13:51:02 -0500
>> From: anthony.minessale at gmail.com
>> To: freeswitch-users at lists.freeswitch.org
>
>> Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database
>> sofia_reg_internal.db
>>
>> I already advised.
>>
>> You did the first one and checked your settings.
>> Next capture sip traffic
>>
>> sofia global siptrace on
>>
>> and look at all the registers for anything anomalous
>>
>> On Wed, Apr 4, 2012 at 1:26 PM, Yuriy Nasida <nasida at live.ru> wrote:
>> > Anthony, thank you for your answer but I think it is not issue with
>> > settings
>> > of sip devices. Incorrect additional row may be any of my customers. I
>> > mean
>> > that the additional row do change after ? ./fs_cli -x "sofia profile
>> > internal flush_inbound_reg"
>> >
>> > Also I have this problem after moving to : FreeSWITCH Version 1.0.head
>> > (git-2ec02e5 2012-03-01 13-22-52 -0600) ?+ Centos 6.
>> >
>> > I really don't understand why i have it just?after cleaning of all
>> > rows...
>> >
>> > sqlite> select * from sip_registrations where sip_user=' user2 ';
>> > | user2 ||||||||||||||||||||
>> >
>> > Please advice.
>> >
>> > Thanks.
>> >
>> > ________________________________
>> > From: nasida at live.ru
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Wed, 4 Apr 2012 21:55:56 +0400
>> > Subject: Re: [Freeswitch-users] incorrect reg rows in sqlite database
>> > sofia_reg_internal.db
>> >
>> >
>> > Update.
>> >
>> > after?./fs_cli -x "sofia profile internal flush_inbound_reg" ?I have
>> >
>> > sqlite> select * from sip_registrations where sip_user=' user2?';
>> > | user2?||||||||||||||||||||
>> >
>> > Why?? I must have clear rows!!
>> >
>> > Next, I will have 2 rows again but with?user3 instead of?user1...
>> >
>> > Thanks.
>> >
>> > ________________________________
>> > From: nasida at live.ru
>> > To: freeswitch-users at lists.freeswitch.org
>> > Date: Wed, 4 Apr 2012 21:20:51 +0400
>> > Subject: [Freeswitch-users] incorrect reg rows in sqlite database
>> > sofia_reg_internal.db
>> >
>> > Hello guys.
>> >
>> > I have issue with incorrect reg rows in sqlite database
>> > sofia_reg_internal.db. it looks like that one registrations are crossed
>> > with
>> > others. so sometimes i have 2 inbound call for 2 different users instead
>> > of
>> > one. What it can be ? probably it is some bug of new FS version ? Can
>> > you
>> > give me some advise ?
>> >
>> > sqlite> select * from sip_registrations where sip_user='user2';
>> >
>> >
>> > 3c73a57f-34be3361 at 10.0.0.140|user2|69.XX.XX.101||??user1?<sip:user1 at 10.0.0.140:5060;fs_nat=yes;fs_path=sip%3Auser1%4079.XX.XX.13%3A5060>|Registered(UDP-NAT)|unknown|1377757948|Linksys/SPA2102-5.2.10;000e081d81c1|user1|69.XX.XX.101|internal|sipproviderdedi1.domain.com|79.XX.XX.13|5060|user1|sip2.sipprovider.com|user1|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>> >
>> > 3bb28227-edc8becf at 80.XX.XX.131|user2|69.XX.XX.101||user2
>> >
>> > <sip:user2 at 80.XX.XX.131:5060>|Registered(UDP)|unknown|1333557925|Linksys/SPA2102-5.2.10;000e081ac13e|user2|69.XX.XX.101|internal|sipproviderdedi1.domain.com|80.XX.XX.131|5060|user2|sip2.sipprovider.com|user2|69.XX.XX.101|69.XX.XX.101|sipproviderdedi1.domain.com|69.XX.XX.101
>> >
>> >
>> > Why I have first row here? ?When I call for?user2 I have two call
>> > (for?user1
>> > an?user2) instead of one (for?user2).
>> > I have separate row for user1...
>> >
>> > Thanks.
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> > CudaTel
>> > Communication Server  Official FreeSWITCH Sites
>> > http://www.freeswitch.org http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> > CudaTel
>> > Communication Server  Official FreeSWITCH Sites
>> > http://www.freeswitch.org http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Apr 13 01:02:10 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Apr 2012 16:02:10 -0500
Subject: [Freeswitch-users] enable_file_write_buffering
In-Reply-To: <CAHCLHGt8dGH9dxkdGbM-gK77SUD16B7XJ5mpMo1QfyeWct8EAQ@mail.gmail.com>
References: <CAHCLHGt8dGH9dxkdGbM-gK77SUD16B7XJ5mpMo1QfyeWct8EAQ@mail.gmail.com>
Message-ID: <CAKbxfG9F+=78580mb33fBZUtq0uELk+i1zm+ciBY0Qmo-fZFtQ@mail.gmail.com>

its raw signed linear audio depending on the sample rate you can
divide by 2 to get samples then divide samples per second.


On Wed, Apr 11, 2012 at 12:00 PM, Tamas Jalsovszky <jalsot at gmail.com> wrote:
> Hello,
>
> Does anybody know how much buffering does the
> enable_file_write_buffering=true provide?
> What I've found from source code that there is a buffer with 65536 length
> but not sure how to calculate the actual buffer length in milliseconds - to
> estimate IOPS at the end.
>
> I would happily add the info to wiki afterwards.
>
> Kind regards,
> ?? Jalsot
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jalsot at gmail.com  Fri Apr 13 01:29:03 2012
From: jalsot at gmail.com (Tamas Jalsovszky)
Date: Thu, 12 Apr 2012 23:29:03 +0200
Subject: [Freeswitch-users] enable_file_write_buffering
In-Reply-To: <CAKbxfG9F+=78580mb33fBZUtq0uELk+i1zm+ciBY0Qmo-fZFtQ@mail.gmail.com>
References: <CAHCLHGt8dGH9dxkdGbM-gK77SUD16B7XJ5mpMo1QfyeWct8EAQ@mail.gmail.com>
	<CAKbxfG9F+=78580mb33fBZUtq0uELk+i1zm+ciBY0Qmo-fZFtQ@mail.gmail.com>
Message-ID: <CAHCLHGtDgpQO2TcXFgStYcBnGLTpCfxn-nyuPNSVdGGvdsPcAw@mail.gmail.com>

Thank you.
Example case: stereo recording, 8kHz, default buffer size (65536)
65536/2 channels/2 bytes perc channel/8000 samples per second -> 2,048
seconds.
Is this a correct calculus?

As I have seen, there is a buffering in media bug too, where the default
number of frames is 25, which would mean in my case with 20ms codec, 500ms
buffer. Does this adds to the one previously calculated meaning the actual
buffering is about 2.5 seconds?

Jalsot

On Thu, Apr 12, 2012 at 11:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its raw signed linear audio depending on the sample rate you can
> divide by 2 to get samples then divide samples per second.
>
>
> On Wed, Apr 11, 2012 at 12:00 PM, Tamas Jalsovszky <jalsot at gmail.com>
> wrote:
> > Hello,
> >
> > Does anybody know how much buffering does the
> > enable_file_write_buffering=true provide?
> > What I've found from source code that there is a buffer with 65536 length
> > but not sure how to calculate the actual buffer length in milliseconds -
> to
> > estimate IOPS at the end.
> >
> > I would happily add the info to wiki afterwards.
> >
> > Kind regards,
> >    Jalsot
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Fri Apr 13 01:46:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Apr 2012 16:46:21 -0500
Subject: [Freeswitch-users] enable_file_write_buffering
In-Reply-To: <CAHCLHGtDgpQO2TcXFgStYcBnGLTpCfxn-nyuPNSVdGGvdsPcAw@mail.gmail.com>
References: <CAHCLHGt8dGH9dxkdGbM-gK77SUD16B7XJ5mpMo1QfyeWct8EAQ@mail.gmail.com>
	<CAKbxfG9F+=78580mb33fBZUtq0uELk+i1zm+ciBY0Qmo-fZFtQ@mail.gmail.com>
	<CAHCLHGtDgpQO2TcXFgStYcBnGLTpCfxn-nyuPNSVdGGvdsPcAw@mail.gmail.com>
Message-ID: <CAKbxfG_4j23gmh+hdzGg6Ge86-7scfJgqTyK2xKeiv9AJyRaGg@mail.gmail.com>

I think that's more or less correct.


On Thu, Apr 12, 2012 at 4:29 PM, Tamas Jalsovszky <jalsot at gmail.com> wrote:
> Thank you.
> Example case: stereo recording, 8kHz, default buffer size (65536)
> 65536/2 channels/2 bytes perc channel/8000 samples per second -> 2,048
> seconds.
> Is this a correct calculus?
>
> As I have seen, there is a buffering in media bug too, where the default
> number of frames is 25, which would mean in my case with 20ms codec, 500ms
> buffer. Does this adds to the one previously calculated meaning the actual
> buffering is about 2.5 seconds?
>
> Jalsot
>
>
> On Thu, Apr 12, 2012 at 11:02 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> its raw signed linear audio depending on the sample rate you can
>> divide by 2 to get samples then divide samples per second.
>>
>>
>> On Wed, Apr 11, 2012 at 12:00 PM, Tamas Jalsovszky <jalsot at gmail.com>
>> wrote:
>> > Hello,
>> >
>> > Does anybody know how much buffering does the
>> > enable_file_write_buffering=true provide?
>> > What I've found from source code that there is a buffer with 65536
>> > length
>> > but not sure how to calculate the actual buffer length in milliseconds -
>> > to
>> > estimate IOPS at the end.
>> >
>> > I would happily add the info to wiki afterwards.
>> >
>> > Kind regards,
>> > ?? Jalsot
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Apr 13 02:39:01 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 12 Apr 2012 17:39:01 -0500
Subject: [Freeswitch-users] Efficient way to query freeswitch - status
	and info
In-Reply-To: <CALn233roTyAEHO_fCmx4XTjCaqVOgrTsDiO=Bsbbk9KnQf3fRw@mail.gmail.com>
References: <CAGvuWYbn6npsd=gSgM88D2MVdS+FPKZoJfv2s2zXYvu0UswoWA@mail.gmail.com>
	<CALn233rxS5BohLWChwKc93679nWr3DERWG01BYgrc+ZGPJmc-g@mail.gmail.com>
	<CALNZuwnsfJ5R_7QerAo3WOvf+UqKkRdTWioBopwfCtm8NZMjMQ@mail.gmail.com>
	<CALn233roTyAEHO_fCmx4XTjCaqVOgrTsDiO=Bsbbk9KnQf3fRw@mail.gmail.com>
Message-ID: <CAKbxfG-AnhVFLY3_4nzCm+YgL8b2n7GaxmFzizmdheNupYOwaA@mail.gmail.com>

You sort of need a little of both cos you need to "bootstrap" the data
with what is already there then update it from there.


On Thu, Apr 12, 2012 at 10:32 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> Many of the status requests now there is an XML option also like
> "sofia xmlstatus" you can use ESL to do that (see wiki for other
> commands). ?Using the events however is better for performance ?to
> avoid polling but does obviously require more logic to maintain state.
>
> ~Mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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From mthakershi at gmail.com  Fri Apr 13 04:20:01 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 12 Apr 2012 19:20:01 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
	<CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
	<CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>
Message-ID: <CAKHK4ZP9PHTJuALOd=JYrzvYO2MttqRn8VYGPTtbsDe1Lzy2tQ@mail.gmail.com>

Thanks.

Before I do what you said, I have another problem. Sorry about that.

I use different mechanism to make outgoing calls (originate calls from
ESL). So I do not use bridge.
I tried to put "{absolute_codec_string=GSM}" in originate call along
with other parameters I pass but it throws error.

My originate call looks like:
originate {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX
1056 XML default

Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION
2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup
sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]

Call without absolute_codec_string works fine.

The way this works is: call gets originated -- it goes to default dial
plan extension 1056 -- mod_managed is called from this point to handle
call flow.

    <extension name="CH1">
	  <!-- Calls from public.xml will come with 1056 extension and will
be handled here -->
      <condition field="destination_number" expression="^1056$">
        <!-- codec trial setting -->
        <!-- <action application="set" data="absolute_codec_string=GSM"/> -->
<!-- 		<action application="log" data="codec debug: codec_string is
${codec_string}" />
		<action application="log" data="codec debug: originator_codec is
${originator_codec}" />
		<action application="log" data="codec debug: absolute_codec_string
is ${absolute_codec_string}" />
 -->		
		<action application="sleep" data="1000" />
        <action application="managed" data="clsAsmtApp" />
      </condition>
    </extension>

As you can see, I tried setting absolute string from here too but it
didn't work.

Let me know what can be done if you can.

Thanks.

Malay


On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
> Lets forget for a minute about?vars.xml?and what is there (when you use
> absolute_codec_string=GSM, it overrides any settings from vars.xml), and
> concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.
>
> Try to change your bridge to the following string:
> <action application="bridge">
> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>
>
> (obviously use your sip provider and phone number :)
>
> And do one outgoing call to this sip provider.
>
> Then, reply back whether the call was successful or not (did you hear the
> voice and whether the other party did hear you). And paste the log to the
> pastebin (http://pastebin.freeswitch.org/) and reply back with the link.
>
>
>
>
>
> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>
>> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>> > use
>> > G729 right now as it needs the license). By OUTGOING I mean call from
>> > FreeSWITCH to your provider, and not back.
>> Yes. I want to try some codec that will use less bandwidth and see how
>> much audio quality is compromised. So I chose GSM because G729 has
>> license requirements.
>>
>> I want this to work when someone calls my DID or my FS program calls
>> someone via DID. So both for incoming calls as well as outgoing calls.
>>
>> > 2. If #1 is true, then?inbound-codec-negotiation is not much useful at
>> > this point. Because this setting is for INCOMING leg, and not for
>> > OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but
>> > still
>> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need
>> > to
>> > find out - why).
>> I am not very familiar with internals of FS so having lot of trouble
>> figuring this out.
>>
>> Weird thing is, even if I do not have any mention of G729, it is being
>> negotiated and used. Here are my configuration lines:
>> vars.xml
>> ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>
>> sip profile has:
>> <param name="inbound-codec-negotiation" value="greedy"/>
>>
>> dial plan
>> ? ? ? ?<extension name="CH">
>> ? ? ?<condition field="destination_number" expression="^105\d$">
>> ? ? ? ? ? ? ? ?<action application="set"
>> data="absolute_codec_string=GSM"/>
>> ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
>> ? ? ? ? ? ? ? ?<action application="managed" data="clsAsmtApp"/>
>> ? ? ?</condition>
>> ? ? ? ? ?</extension>
>>
>>
>> ?<extension name="vitel-outbound">
>> ? <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>> ? ? ? ? <action application="set" data="absolute_codec_string=GSM"/>
>> ? ? <action application="set"
>> data="effective_caller_id_number=18770001111"/>
>> ? ? <action application="bridge"
>> data="sofia/gateway/vitelity-outbound/$1"/>
>> ? ?</condition>
>> ?</extension>
>>
>> > 3. Try using absolute_codec_string before bridging your call to your SIP
>> > provider:
>> >
>> > <action application="set" data="absolute_codec_string=GSM"/>
>> > <action application="bridge"
>> > data="sofia/gateway/your_provider/12345678901_example"/>
>> >
>> I did as you said. But found what I described above. If I remove
>> "greedy" setting, system simply uses PCMU codec like before.
>>
>> This thing is so confusing. Even my SIP provider is saying they can't
>> help out with FS because some guy quit who knew the configurations.
>>
>> Thanks for help.
>>
>> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
>> wrote:
>> >
>> > Lets confirm the following statements now:
>> >
>> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>> > use
>> > G729 right now as it needs the license). By OUTGOING I mean call from
>> > FreeSWITCH to your provider, and not back.
>> >
>> > 2. If #1 is true, then?inbound-codec-negotiation is not much useful at
>> > this point. Because this setting is for INCOMING leg, and not for
>> > OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but
>> > still
>> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need
>> > to
>> > find out - why).
>> >
>> > 3. Try using absolute_codec_string before bridging your call to your SIP
>> > provider:
>> >
>> > <action application="set" data="absolute_codec_string=GSM"/>
>> > <action application="bridge"
>> > data="sofia/gateway/your_provider/12345678901_example"/>
>> >
>> >
>> > And reply back if this changes anything...
>> >
>> > Vitalie
>> >
>> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>> >>
>> >> Under my sipinterface_1 profile that I use for these calls, I have:
>> >>
>> >> <param name="inbound-codec-negotiation" value="greedy"/>
>> >>
>> >> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>> >>>
>> >>> On Wed, 11 Apr 2012 13:31:45 -0500
>> >>> Malay Thakershi <mthakershi at gmail.com> wrote:
>> >>>
>> >>> > Understood your point about G722. But if my list has GSM at higher
>> >>> > priority
>> >>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the call
>> >>> > using
>> >>> > GSM?
>> >>> >
>> >>> > I don't know where G726 comparisons are coming from. If I haven't
>> >>> > mentioned
>> >>> > it in vars.xml and far-end SDP doesn't support it then why would it
>> >>> > even
>> >>> > consider that codec?
>> >>>
>> >>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >>> (inbound-codec-negotiation):
>> >>> 'generous' permits the remote codec list have precedence and 'win'
>> >>> ? the codec negotiation and selection process
>> >>>
>> >>> JY
>> >>> --
>> >>> Don't drop acid -- take it pass/fail.
>> >>> ? ? ? ? ? ? ? ?-- Seen in a Ladies' Room at Harvard
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From vetali100 at gmail.com  Fri Apr 13 07:55:00 2012
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 12 Apr 2012 20:55:00 -0700
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZP9PHTJuALOd=JYrzvYO2MttqRn8VYGPTtbsDe1Lzy2tQ@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
	<CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
	<CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>
	<CAKHK4ZP9PHTJuALOd=JYrzvYO2MttqRn8VYGPTtbsDe1Lzy2tQ@mail.gmail.com>
Message-ID: <CACc3D0XRc762wzCUdvxfjbo69Q0v6cMpMsQSwVhBn=_z-gmbXg@mail.gmail.com>

Could you collect the full siptrace of one call and paste into pastebin?

To do so, type at the cli:

       sofia global siptrace on

Then make 1 call and collect whatever you got at the screen, and paste into
pastebin.

Then, you'll probably want to disable siptrace:

        sofia global siptrace off

Regards,
Vitalie


2012/4/12 Malay Thakershi <mthakershi at gmail.com>

> Thanks.
>
> Before I do what you said, I have another problem. Sorry about that.
>
> I use different mechanism to make outgoing calls (originate calls from
> ESL). So I do not use bridge.
> I tried to put "{absolute_codec_string=GSM}" in originate call along
> with other parameters I pass but it throws error.
>
> My originate call looks like:
> originate
> {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX
> 1056 XML default
>
> Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION
> 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup
> sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]
> [INCOMPATIBLE_DESTINATION]
>
> Call without absolute_codec_string works fine.
>
> The way this works is: call gets originated -- it goes to default dial
> plan extension 1056 -- mod_managed is called from this point to handle
> call flow.
>
>    <extension name="CH1">
>          <!-- Calls from public.xml will come with 1056 extension and will
> be handled here -->
>      <condition field="destination_number" expression="^1056$">
>        <!-- codec trial setting -->
>        <!-- <action application="set" data="absolute_codec_string=GSM"/>
> -->
> <!--            <action application="log" data="codec debug: codec_string
> is
> ${codec_string}" />
>                <action application="log" data="codec debug:
> originator_codec is
> ${originator_codec}" />
>                <action application="log" data="codec debug:
> absolute_codec_string
> is ${absolute_codec_string}" />
>  -->
>                 <action application="sleep" data="1000" />
>        <action application="managed" data="clsAsmtApp" />
>      </condition>
>    </extension>
>
> As you can see, I tried setting absolute string from here too but it
> didn't work.
>
> Let me know what can be done if you can.
>
> Thanks.
>
> Malay
>
>
> On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100 at gmail.com>
> wrote:
> > Lets forget for a minute about vars.xml and what is there (when you use
> > absolute_codec_string=GSM, it overrides any settings from vars.xml), and
> > concentrate at the OUTGOING call from FreeSWITCH to your sip provider
> first.
> >
> > Try to change your bridge to the following string:
> > <action application="bridge">
> >
> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>
> >
> > (obviously use your sip provider and phone number :)
> >
> > And do one outgoing call to this sip provider.
> >
> > Then, reply back whether the call was successful or not (did you hear the
> > voice and whether the other party did hear you). And paste the log to the
> > pastebin (http://pastebin.freeswitch.org/) and reply back with the link.
> >
> >
> >
> >
> >
> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
> >>
> >> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
> >> > use
> >> > G729 right now as it needs the license). By OUTGOING I mean call from
> >> > FreeSWITCH to your provider, and not back.
> >> Yes. I want to try some codec that will use less bandwidth and see how
> >> much audio quality is compromised. So I chose GSM because G729 has
> >> license requirements.
> >>
> >> I want this to work when someone calls my DID or my FS program calls
> >> someone via DID. So both for incoming calls as well as outgoing calls.
> >>
> >> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at
> >> > this point. Because this setting is for INCOMING leg, and not for
> >> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
> >> > still
> >> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
> need
> >> > to
> >> > find out - why).
> >> I am not very familiar with internals of FS so having lot of trouble
> >> figuring this out.
> >>
> >> Weird thing is, even if I do not have any mention of G729, it is being
> >> negotiated and used. Here are my configuration lines:
> >> vars.xml
> >>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
> >>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
> >>
> >> sip profile has:
> >> <param name="inbound-codec-negotiation" value="greedy"/>
> >>
> >> dial plan
> >>        <extension name="CH">
> >>      <condition field="destination_number" expression="^105\d$">
> >>                <action application="set"
> >> data="absolute_codec_string=GSM"/>
> >>                <action application="sleep" data="1000"/>
> >>                <action application="managed" data="clsAsmtApp"/>
> >>      </condition>
> >>          </extension>
> >>
> >>
> >>  <extension name="vitel-outbound">
> >>   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
> >>         <action application="set" data="absolute_codec_string=GSM"/>
> >>     <action application="set"
> >> data="effective_caller_id_number=18770001111"/>
> >>     <action application="bridge"
> >> data="sofia/gateway/vitelity-outbound/$1"/>
> >>    </condition>
> >>  </extension>
> >>
> >> > 3. Try using absolute_codec_string before bridging your call to your
> SIP
> >> > provider:
> >> >
> >> > <action application="set" data="absolute_codec_string=GSM"/>
> >> > <action application="bridge"
> >> > data="sofia/gateway/your_provider/12345678901_example"/>
> >> >
> >> I did as you said. But found what I described above. If I remove
> >> "greedy" setting, system simply uses PCMU codec like before.
> >>
> >> This thing is so confusing. Even my SIP provider is saying they can't
> >> help out with FS because some guy quit who knew the configurations.
> >>
> >> Thanks for help.
> >>
> >> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
> >> wrote:
> >> >
> >> > Lets confirm the following statements now:
> >> >
> >> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
> >> > use
> >> > G729 right now as it needs the license). By OUTGOING I mean call from
> >> > FreeSWITCH to your provider, and not back.
> >> >
> >> > 2. If #1 is true, then inbound-codec-negotiation is not much useful at
> >> > this point. Because this setting is for INCOMING leg, and not for
> >> > OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
> >> > still
> >> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
> need
> >> > to
> >> > find out - why).
> >> >
> >> > 3. Try using absolute_codec_string before bridging your call to your
> SIP
> >> > provider:
> >> >
> >> > <action application="set" data="absolute_codec_string=GSM"/>
> >> > <action application="bridge"
> >> > data="sofia/gateway/your_provider/12345678901_example"/>
> >> >
> >> >
> >> > And reply back if this changes anything...
> >> >
> >> > Vitalie
> >> >
> >> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
> >> >>
> >> >> Under my sipinterface_1 profile that I use for these calls, I have:
> >> >>
> >> >> <param name="inbound-codec-negotiation" value="greedy"/>
> >> >>
> >> >> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
> >> >>>
> >> >>> On Wed, 11 Apr 2012 13:31:45 -0500
> >> >>> Malay Thakershi <mthakershi at gmail.com> wrote:
> >> >>>
> >> >>> > Understood your point about G722. But if my list has GSM at higher
> >> >>> > priority
> >> >>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the
> call
> >> >>> > using
> >> >>> > GSM?
> >> >>> >
> >> >>> > I don't know where G726 comparisons are coming from. If I haven't
> >> >>> > mentioned
> >> >>> > it in vars.xml and far-end SDP doesn't support it then why would
> it
> >> >>> > even
> >> >>> > consider that codec?
> >> >>>
> >> >>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
> >> >>> (inbound-codec-negotiation):
> >> >>> 'generous' permits the remote codec list have precedence and 'win'
> >> >>>   the codec negotiation and selection process
> >> >>>
> >> >>> JY
> >> >>> --
> >> >>> Don't drop acid -- take it pass/fail.
> >> >>>                -- Seen in a Ladies' Room at Harvard
> >> >>>
> >> >>>
> >> >>>
> >> >>>
> _________________________________________________________________________
> >> >>> Professional FreeSWITCH Consulting Services:
> >> >>> consulting at freeswitch.org
> >> >>> http://www.freeswitchsolutions.com
> >> >>>
> >> >>> 
> >> >>> 
> >> >>>
> >> >>> Official FreeSWITCH Sites
> >> >>> http://www.freeswitch.org
> >> >>> http://wiki.freeswitch.org
> >> >>> http://www.cluecon.com
> >> >>>
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >>
> >> >>
> >> >>
> >> >>
> _________________________________________________________________________
> >> >> Professional FreeSWITCH Consulting Services:
> >> >> consulting at freeswitch.org
> >> >> http://www.freeswitchsolutions.com
> >> >>
> >> >> 
> >> >> 
> >> >>
> >> >> Official FreeSWITCH Sites
> >> >> http://www.freeswitch.org
> >> >> http://wiki.freeswitch.org
> >> >> http://www.cluecon.com
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mthakershi at gmail.com  Fri Apr 13 09:39:54 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Fri, 13 Apr 2012 00:39:54 -0500
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CACc3D0XRc762wzCUdvxfjbo69Q0v6cMpMsQSwVhBn=_z-gmbXg@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
	<CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
	<CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>
	<CAKHK4ZP9PHTJuALOd=JYrzvYO2MttqRn8VYGPTtbsDe1Lzy2tQ@mail.gmail.com>
	<CACc3D0XRc762wzCUdvxfjbo69Q0v6cMpMsQSwVhBn=_z-gmbXg@mail.gmail.com>
Message-ID: <CAKHK4ZNTDhB6t=tirRqgi95ZKCuFoe-JFYYyWEL8Nt4zbr2+kw@mail.gmail.com>

When I do "sofia global siptrace on", my console just doesn't stop. I
get text so fast, I can't even paste the originate command. So I can't
produce what will hep you help me on this.

Is there a way I can divert the whole thing to a text file?

Thanks.

Malay

On Thu, Apr 12, 2012 at 10:55 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
> Could you collect the full siptrace of one call and paste into pastebin?
>
> To do so, type at the cli:
>
> ? ? ? ?sofia global siptrace on
>
> Then make 1 call and collect whatever you got at the screen, and paste into
> pastebin.
>
> Then, you'll probably want to disable siptrace:
>
> ? ? ? ? sofia global siptrace off
>
> Regards,
> Vitalie
>
>
> 2012/4/12 Malay Thakershi <mthakershi at gmail.com>
>>
>> Thanks.
>>
>> Before I do what you said, I have another problem. Sorry about that.
>>
>> I use different mechanism to make outgoing calls (originate calls from
>> ESL). So I do not use bridge.
>> I tried to put "{absolute_codec_string=GSM}" in originate call along
>> with other parameters I pass but it throws error.
>>
>> My originate call looks like:
>> originate
>> {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX
>> 1056 XML default
>>
>> Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION
>> 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup
>> sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]
>> [INCOMPATIBLE_DESTINATION]
>>
>> Call without absolute_codec_string works fine.
>>
>> The way this works is: call gets originated -- it goes to default dial
>> plan extension 1056 -- mod_managed is called from this point to handle
>> call flow.
>>
>> ? ?<extension name="CH1">
>> ? ? ? ? ?<!-- Calls from public.xml will come with 1056 extension and will
>> be handled here -->
>> ? ? ?<condition field="destination_number" expression="^1056$">
>> ? ? ? ?<!-- codec trial setting -->
>> ? ? ? ?<!-- <action application="set" data="absolute_codec_string=GSM"/>
>> -->
>> <!-- ? ? ? ? ? ?<action application="log" data="codec debug: codec_string
>> is
>> ${codec_string}" />
>> ? ? ? ? ? ? ? ?<action application="log" data="codec debug:
>> originator_codec is
>> ${originator_codec}" />
>> ? ? ? ? ? ? ? ?<action application="log" data="codec debug:
>> absolute_codec_string
>> is ${absolute_codec_string}" />
>> ?-->
>> ? ? ? ? ? ? ? ?<action application="sleep" data="1000" />
>> ? ? ? ?<action application="managed" data="clsAsmtApp" />
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> As you can see, I tried setting absolute string from here too but it
>> didn't work.
>>
>> Let me know what can be done if you can.
>>
>> Thanks.
>>
>> Malay
>>
>>
>> On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100 at gmail.com>
>> wrote:
>> > Lets forget for a minute about?vars.xml?and what is there (when you use
>> > absolute_codec_string=GSM, it overrides any settings from vars.xml), and
>> > concentrate at the OUTGOING call from FreeSWITCH to your sip provider
>> > first.
>> >
>> > Try to change your bridge to the following string:
>> > <action application="bridge">
>> >
>> > data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>
>> >
>> > (obviously use your sip provider and phone number :)
>> >
>> > And do one outgoing call to this sip provider.
>> >
>> > Then, reply back whether the call was successful or not (did you hear
>> > the
>> > voice and whether the other party did hear you). And paste the log to
>> > the
>> > pastebin (http://pastebin.freeswitch.org/) and reply back with the link.
>> >
>> >
>> >
>> >
>> >
>> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>> >>
>> >> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>> >> > use
>> >> > G729 right now as it needs the license). By OUTGOING I mean call from
>> >> > FreeSWITCH to your provider, and not back.
>> >> Yes. I want to try some codec that will use less bandwidth and see how
>> >> much audio quality is compromised. So I chose GSM because G729 has
>> >> license requirements.
>> >>
>> >> I want this to work when someone calls my DID or my FS program calls
>> >> someone via DID. So both for incoming calls as well as outgoing calls.
>> >>
>> >> > 2. If #1 is true, then?inbound-codec-negotiation is not much useful
>> >> > at
>> >> > this point. Because this setting is for INCOMING leg, and not for
>> >> > OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but
>> >> > still
>> >> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>> >> > need
>> >> > to
>> >> > find out - why).
>> >> I am not very familiar with internals of FS so having lot of trouble
>> >> figuring this out.
>> >>
>> >> Weird thing is, even if I do not have any mention of G729, it is being
>> >> negotiated and used. Here are my configuration lines:
>> >> vars.xml
>> >> ?<X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>> >> ?<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>> >>
>> >> sip profile has:
>> >> <param name="inbound-codec-negotiation" value="greedy"/>
>> >>
>> >> dial plan
>> >> ? ? ? ?<extension name="CH">
>> >> ? ? ?<condition field="destination_number" expression="^105\d$">
>> >> ? ? ? ? ? ? ? ?<action application="set"
>> >> data="absolute_codec_string=GSM"/>
>> >> ? ? ? ? ? ? ? ?<action application="sleep" data="1000"/>
>> >> ? ? ? ? ? ? ? ?<action application="managed" data="clsAsmtApp"/>
>> >> ? ? ?</condition>
>> >> ? ? ? ? ?</extension>
>> >>
>> >>
>> >> ?<extension name="vitel-outbound">
>> >> ? <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>> >> ? ? ? ? <action application="set" data="absolute_codec_string=GSM"/>
>> >> ? ? <action application="set"
>> >> data="effective_caller_id_number=18770001111"/>
>> >> ? ? <action application="bridge"
>> >> data="sofia/gateway/vitelity-outbound/$1"/>
>> >> ? ?</condition>
>> >> ?</extension>
>> >>
>> >> > 3. Try using absolute_codec_string before bridging your call to your
>> >> > SIP
>> >> > provider:
>> >> >
>> >> > <action application="set" data="absolute_codec_string=GSM"/>
>> >> > <action application="bridge"
>> >> > data="sofia/gateway/your_provider/12345678901_example"/>
>> >> >
>> >> I did as you said. But found what I described above. If I remove
>> >> "greedy" setting, system simply uses PCMU codec like before.
>> >>
>> >> This thing is so confusing. Even my SIP provider is saying they can't
>> >> help out with FS because some guy quit who knew the configurations.
>> >>
>> >> Thanks for help.
>> >>
>> >> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
>> >> wrote:
>> >> >
>> >> > Lets confirm the following statements now:
>> >> >
>> >> > 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>> >> > use
>> >> > G729 right now as it needs the license). By OUTGOING I mean call from
>> >> > FreeSWITCH to your provider, and not back.
>> >> >
>> >> > 2. If #1 is true, then?inbound-codec-negotiation is not much useful
>> >> > at
>> >> > this point. Because this setting is for INCOMING leg, and not for
>> >> > OUTGOING.?And even if it _might_ be using GSM for INCOMING leg, but
>> >> > still
>> >> > transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>> >> > need
>> >> > to
>> >> > find out - why).
>> >> >
>> >> > 3. Try using absolute_codec_string before bridging your call to your
>> >> > SIP
>> >> > provider:
>> >> >
>> >> > <action application="set" data="absolute_codec_string=GSM"/>
>> >> > <action application="bridge"
>> >> > data="sofia/gateway/your_provider/12345678901_example"/>
>> >> >
>> >> >
>> >> > And reply back if this changes anything...
>> >> >
>> >> > Vitalie
>> >> >
>> >> > 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>> >> >>
>> >> >> Under my sipinterface_1 profile that I use for these calls, I have:
>> >> >>
>> >> >> <param name="inbound-codec-negotiation" value="greedy"/>
>> >> >>
>> >> >> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>> >> >>>
>> >> >>> On Wed, 11 Apr 2012 13:31:45 -0500
>> >> >>> Malay Thakershi <mthakershi at gmail.com> wrote:
>> >> >>>
>> >> >>> > Understood your point about G722. But if my list has GSM at
>> >> >>> > higher
>> >> >>> > priority
>> >> >>> > over PCMU/PCMA and far-end SDP supports GSM then why isn't the
>> >> >>> > call
>> >> >>> > using
>> >> >>> > GSM?
>> >> >>> >
>> >> >>> > I don't know where G726 comparisons are coming from. If I haven't
>> >> >>> > mentioned
>> >> >>> > it in vars.xml and far-end SDP doesn't support it then why would
>> >> >>> > it
>> >> >>> > even
>> >> >>> > consider that codec?
>> >> >>>
>> >> >>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >> >>> (inbound-codec-negotiation):
>> >> >>> 'generous' permits the remote codec list have precedence and 'win'
>> >> >>> ? the codec negotiation and selection process
>> >> >>>
>> >> >>> JY
>> >> >>> --
>> >> >>> Don't drop acid -- take it pass/fail.
>> >> >>> ? ? ? ? ? ? ? ?-- Seen in a Ladies' Room at Harvard
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>>
>> >> >>> _________________________________________________________________________
>> >> >>> Professional FreeSWITCH Consulting Services:
>> >> >>> consulting at freeswitch.org
>> >> >>> http://www.freeswitchsolutions.com
>> >> >>>
>> >> >>> 
>> >> >>> 
>> >> >>>
>> >> >>> Official FreeSWITCH Sites
>> >> >>> http://www.freeswitch.org
>> >> >>> http://wiki.freeswitch.org
>> >> >>> http://www.cluecon.com
>> >> >>>
>> >> >>> FreeSWITCH-users mailing list
>> >> >>> FreeSWITCH-users at lists.freeswitch.org
>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>>
>> >> >>>
>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >>> http://www.freeswitch.org
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> _________________________________________________________________________
>> >> >> Professional FreeSWITCH Consulting Services:
>> >> >> consulting at freeswitch.org
>> >> >> http://www.freeswitchsolutions.com
>> >> >>
>> >> >> 
>> >> >> 
>> >> >>
>> >> >> Official FreeSWITCH Sites
>> >> >> http://www.freeswitch.org
>> >> >> http://wiki.freeswitch.org
>> >> >> http://www.cluecon.com
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >>
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > _________________________________________________________________________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > 
>> >> > 
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://wiki.freeswitch.org
>> >> > http://www.cluecon.com
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Fri Apr 13 10:06:59 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 13 Apr 2012 06:06:59 +0000
Subject: [Freeswitch-users] Possible to disable core codecs?
In-Reply-To: <CAKHK4ZNTDhB6t=tirRqgi95ZKCuFoe-JFYYyWEL8Nt4zbr2+kw@mail.gmail.com>
References: <CAKHK4ZMftjPNB1a=3XeMSsmdRc+6qPwu6NGMKMT9yHT5JZh2Nw@mail.gmail.com>
	<CAKzWOxW2Mf=qCJt6MwBQUMq0Dk-RdEgL=qO6C9+Z8Ezr9PaX8g@mail.gmail.com>
	<CALNZuwmss9YSNBebwSWNuTtF+gRsfZp+7O3bwy6nHKur6vy9Lw@mail.gmail.com>
	<CAKHK4ZOdJsPfodBvsOjzq4oww6BPXgGfg+JZS5u_ajUuTQcOXQ@mail.gmail.com>
	<1FFF97C269757C458224B7C895F35F1508CABC@cantor.std.visionutv.se>
	<CAKHK4ZNcMu3e0gWzNZJN1mwh=aD=PHRobWhkjpUwkQeBQuSCUg@mail.gmail.com>
	<CAKV-jFyh+MF=qA+h25sHJS34tfC1innjtUMbAd6RyJsJ=6zzRA@mail.gmail.com>
	<CAKzWOxX33QYaQZsXyEfTbR9Mg_M4mvcmdKdQ2hX_aa=+R3qk7w@mail.gmail.com>
	<4F848866.5010008@coppice.org>
	<CAKzWOxWupTtYRyELvi2QO3KFqHiiAW_iksNa5dMez4ybtn4bUA@mail.gmail.com>
	<CAKHK4ZPSX5dus04zv11HJyGb=+g5EMevSRy9tpVXEuM3PcBpLQ@mail.gmail.com>
	<4F85C524.9070106@cupis.co.uk>
	<CAKHK4ZPdGUSUfuMKd9kAqh8uBWvVCbqcN_r53Q=bid5aJDk3eA@mail.gmail.com>
	<20120411204331.0e8017fa@anubis.defcon1>
	<CAKHK4ZNViLmuyb0AiHmKDnRJ53Xrz0JQJYn4WRCS2kcF7-gKzQ@mail.gmail.com>
	<CACc3D0W+Z8jy=eeGyThzNWGD+SBJVnMTnKFf1jKH8KSdXLuSPg@mail.gmail.com>
	<CAKHK4ZP7gdrtNpuGSg1vmtYaZ6_s4vnE0umU9CCpNDFX6khqww@mail.gmail.com>
	<CACc3D0XO6v6GN-bvWOePiuKhFA4WL5PdhPBT_rQntM_VAF_pog@mail.gmail.com>
	<CAKHK4ZP9PHTJuALOd=JYrzvYO2MttqRn8VYGPTtbsDe1Lzy2tQ@mail.gmail.com>
	<CACc3D0XRc762wzCUdvxfjbo69Q0v6cMpMsQSwVhBn=_z-gmbXg@mail.gmail.com>,
	<CAKHK4ZNTDhB6t=tirRqgi95ZKCuFoe-JFYYyWEL8Nt4zbr2+kw@mail.gmail.com>
Message-ID: <E01A9C52-560E-4555-A5D3-457E650B8CA3@visionutveckling.se>

Just pipe it to a file. fs_cli > file.txt.

However, without seeing any logs there are really only two options, the remote node doesn't offer GSM codec, or GSM codec is not loaded in FS (IIRC mod_spandsp handles this codec).

/Peter

13 apr 2012 kl. 07:45 skrev "Malay Thakershi" <mthakershi at gmail.com>:

> When I do "sofia global siptrace on", my console just doesn't stop. I
> get text so fast, I can't even paste the originate command. So I can't
> produce what will hep you help me on this.
> 
> Is there a way I can divert the whole thing to a text file?
> 
> Thanks.
> 
> Malay
> 
> On Thu, Apr 12, 2012 at 10:55 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
>> Could you collect the full siptrace of one call and paste into pastebin?
>> 
>> To do so, type at the cli:
>> 
>>        sofia global siptrace on
>> 
>> Then make 1 call and collect whatever you got at the screen, and paste into
>> pastebin.
>> 
>> Then, you'll probably want to disable siptrace:
>> 
>>         sofia global siptrace off
>> 
>> Regards,
>> Vitalie
>> 
>> 
>> 2012/4/12 Malay Thakershi <mthakershi at gmail.com>
>>> 
>>> Thanks.
>>> 
>>> Before I do what you said, I have another problem. Sorry about that.
>>> 
>>> I use different mechanism to make outgoing calls (originate calls from
>>> ESL). So I do not use bridge.
>>> I tried to put "{absolute_codec_string=GSM}" in originate call along
>>> with other parameters I pass but it throws error.
>>> 
>>> My originate call looks like:
>>> originate
>>> {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX
>>> 1056 XML default
>>> 
>>> Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION
>>> 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup
>>> sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA]
>>> [INCOMPATIBLE_DESTINATION]
>>> 
>>> Call without absolute_codec_string works fine.
>>> 
>>> The way this works is: call gets originated -- it goes to default dial
>>> plan extension 1056 -- mod_managed is called from this point to handle
>>> call flow.
>>> 
>>>    <extension name="CH1">
>>>          <!-- Calls from public.xml will come with 1056 extension and will
>>> be handled here -->
>>>      <condition field="destination_number" expression="^1056$">
>>>        <!-- codec trial setting -->
>>>        <!-- <action application="set" data="absolute_codec_string=GSM"/>
>>> -->
>>> <!--            <action application="log" data="codec debug: codec_string
>>> is
>>> ${codec_string}" />
>>>                <action application="log" data="codec debug:
>>> originator_codec is
>>> ${originator_codec}" />
>>>                <action application="log" data="codec debug:
>>> absolute_codec_string
>>> is ${absolute_codec_string}" />
>>>  -->
>>>                <action application="sleep" data="1000" />
>>>        <action application="managed" data="clsAsmtApp" />
>>>      </condition>
>>>    </extension>
>>> 
>>> As you can see, I tried setting absolute string from here too but it
>>> didn't work.
>>> 
>>> Let me know what can be done if you can.
>>> 
>>> Thanks.
>>> 
>>> Malay
>>> 
>>> 
>>> On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100 at gmail.com>
>>> wrote:
>>>> Lets forget for a minute about vars.xml and what is there (when you use
>>>> absolute_codec_string=GSM, it overrides any settings from vars.xml), and
>>>> concentrate at the OUTGOING call from FreeSWITCH to your sip provider
>>>> first.
>>>> 
>>>> Try to change your bridge to the following string:
>>>> <action application="bridge">
>>>> 
>>>> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>
>>>> 
>>>> (obviously use your sip provider and phone number :)
>>>> 
>>>> And do one outgoing call to this sip provider.
>>>> 
>>>> Then, reply back whether the call was successful or not (did you hear
>>>> the
>>>> voice and whether the other party did hear you). And paste the log to
>>>> the
>>>> pastebin (http://pastebin.freeswitch.org/) and reply back with the link.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>>>> 
>>>>>> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>>>>>> use
>>>>>> G729 right now as it needs the license). By OUTGOING I mean call from
>>>>>> FreeSWITCH to your provider, and not back.
>>>>> Yes. I want to try some codec that will use less bandwidth and see how
>>>>> much audio quality is compromised. So I chose GSM because G729 has
>>>>> license requirements.
>>>>> 
>>>>> I want this to work when someone calls my DID or my FS program calls
>>>>> someone via DID. So both for incoming calls as well as outgoing calls.
>>>>> 
>>>>>> 2. If #1 is true, then inbound-codec-negotiation is not much useful
>>>>>> at
>>>>>> this point. Because this setting is for INCOMING leg, and not for
>>>>>> OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
>>>>>> still
>>>>>> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>>>>>> need
>>>>>> to
>>>>>> find out - why).
>>>>> I am not very familiar with internals of FS so having lot of trouble
>>>>> figuring this out.
>>>>> 
>>>>> Weird thing is, even if I do not have any mention of G729, it is being
>>>>> negotiated and used. Here are my configuration lines:
>>>>> vars.xml
>>>>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>
>>>>>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>
>>>>> 
>>>>> sip profile has:
>>>>> <param name="inbound-codec-negotiation" value="greedy"/>
>>>>> 
>>>>> dial plan
>>>>>        <extension name="CH">
>>>>>      <condition field="destination_number" expression="^105\d$">
>>>>>                <action application="set"
>>>>> data="absolute_codec_string=GSM"/>
>>>>>                <action application="sleep" data="1000"/>
>>>>>                <action application="managed" data="clsAsmtApp"/>
>>>>>      </condition>
>>>>>          </extension>
>>>>> 
>>>>> 
>>>>>  <extension name="vitel-outbound">
>>>>>   <condition field="destination_number" expression="^(1{0,1}\d{10})$">
>>>>>         <action application="set" data="absolute_codec_string=GSM"/>
>>>>>     <action application="set"
>>>>> data="effective_caller_id_number=18770001111"/>
>>>>>     <action application="bridge"
>>>>> data="sofia/gateway/vitelity-outbound/$1"/>
>>>>>    </condition>
>>>>>  </extension>
>>>>> 
>>>>>> 3. Try using absolute_codec_string before bridging your call to your
>>>>>> SIP
>>>>>> provider:
>>>>>> 
>>>>>> <action application="set" data="absolute_codec_string=GSM"/>
>>>>>> <action application="bridge"
>>>>>> data="sofia/gateway/your_provider/12345678901_example"/>
>>>>>> 
>>>>> I did as you said. But found what I described above. If I remove
>>>>> "greedy" setting, system simply uses PCMU codec like before.
>>>>> 
>>>>> This thing is so confusing. Even my SIP provider is saying they can't
>>>>> help out with FS because some guy quit who knew the configurations.
>>>>> 
>>>>> Thanks for help.
>>>>> 
>>>>> On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100 at gmail.com>
>>>>> wrote:
>>>>>> 
>>>>>> Lets confirm the following statements now:
>>>>>> 
>>>>>> 1.You need OUTGOING leg to use GSM codec, right? (You wont be able to
>>>>>> use
>>>>>> G729 right now as it needs the license). By OUTGOING I mean call from
>>>>>> FreeSWITCH to your provider, and not back.
>>>>>> 
>>>>>> 2. If #1 is true, then inbound-codec-negotiation is not much useful
>>>>>> at
>>>>>> this point. Because this setting is for INCOMING leg, and not for
>>>>>> OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but
>>>>>> still
>>>>>> transcode to the OUTGOING PCMA/PCMU by some reason (this is what we
>>>>>> need
>>>>>> to
>>>>>> find out - why).
>>>>>> 
>>>>>> 3. Try using absolute_codec_string before bridging your call to your
>>>>>> SIP
>>>>>> provider:
>>>>>> 
>>>>>> <action application="set" data="absolute_codec_string=GSM"/>
>>>>>> <action application="bridge"
>>>>>> data="sofia/gateway/your_provider/12345678901_example"/>
>>>>>> 
>>>>>> 
>>>>>> And reply back if this changes anything...
>>>>>> 
>>>>>> Vitalie
>>>>>> 
>>>>>> 2012/4/11 Malay Thakershi <mthakershi at gmail.com>
>>>>>>> 
>>>>>>> Under my sipinterface_1 profile that I use for these calls, I have:
>>>>>>> 
>>>>>>> <param name="inbound-codec-negotiation" value="greedy"/>
>>>>>>> 
>>>>>>> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus at gmx.com> wrote:
>>>>>>>> 
>>>>>>>> On Wed, 11 Apr 2012 13:31:45 -0500
>>>>>>>> Malay Thakershi <mthakershi at gmail.com> wrote:
>>>>>>>> 
>>>>>>>>> Understood your point about G722. But if my list has GSM at
>>>>>>>>> higher
>>>>>>>>> priority
>>>>>>>>> over PCMU/PCMA and far-end SDP supports GSM then why isn't the
>>>>>>>>> call
>>>>>>>>> using
>>>>>>>>> GSM?
>>>>>>>>> 
>>>>>>>>> I don't know where G726 comparisons are coming from. If I haven't
>>>>>>>>> mentioned
>>>>>>>>> it in vars.xml and far-end SDP doesn't support it then why would
>>>>>>>>> it
>>>>>>>>> even
>>>>>>>>> consider that codec?
>>>>>>>> 
>>>>>>>> From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>>>>>>> (inbound-codec-negotiation):
>>>>>>>> 'generous' permits the remote codec list have precedence and 'win'
>>>>>>>>   the codec negotiation and selection process
>>>>>>>> 
>>>>>>>> JY
>>>>>>>> --
>>>>>>>> Don't drop acid -- take it pass/fail.
>>>>>>>>                -- Seen in a Ladies' Room at Harvard
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> 
>>>>>>>> 
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4f87bae732768532145870!
> 


From ocset at the800group.com  Fri Apr 13 10:33:42 2012
From: ocset at the800group.com (ocset)
Date: Fri, 13 Apr 2012 14:33:42 +0800
Subject: [Freeswitch-users] Problem bridging to an external mobile number
In-Reply-To: <CAKnNFz-sF2xT5HgRROVG6eiYGUwqGaK_PK5r2q8FvyRDV9kiuw@mail.gmail.com>
References: <4F83C305.7070606@the800group.com>
	<CAKnNFz-sF2xT5HgRROVG6eiYGUwqGaK_PK5r2q8FvyRDV9kiuw@mail.gmail.com>
Message-ID: <4F87C8C6.1020503@the800group.com>

I think I have identifies the culprit.

I'm have a VOIP line with iinet and I read somewhere that they set 
"Max-Forwards=4", which may be causing an issue (any one confirm this?). 
I registered with a different provider (pennytel) and the bridge works. 
I have had to remove the GSM gateway out of the equation due to the 
power supply failing on me so I am hoping that does not have anything to 
do with the problem going away.

Thanks
O.

On 04/12/2012 05:01 AM, chris wrote:
> what gsm gateway are you using? ive been looking for a good one
>
> On Tue, Apr 10, 2012 at 1:20 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     My configuration is a GXW4104 (IP 192.168.0.160) with two PSTN lines.
>     The first line is a POTS line and the second line is a GSM gateway
>     device. Both devices work correctly if I dial out but when I try and
>     bridge between the two, nothing works as expected. The bridge is from
>     the POTS line to a mobile number using the GSM gateway.
>
>     I have created the following bridge for the POTS line (phone numbers
>     removed for privacy reasons)
>
>     <extension name="pots_line">
>     <condition field="destination_number" expression="^(xxxxxxxxxx)$">
>     <action application="bridge"
>     data="sofia/internal/zzzzzzzzzz at 192.168.0.160:5062
>     <http://zzzzzzzzzz at 192.168.0.160:5062>"/>
>     </condition>
>     </extension>
>
>     The bridge does eventually happen (long delay) but neither party can
>     hear each other. Also, when I hang-up with one or both phones, the
>     connection does not terminate and I have to reboot the GXW4104 to
>     break
>     the connection.
>
>     I have bought the Freeswitch books and this seems to be something that
>     should easily work. I have not added any variables into the
>     <extension>
>     definition as this is a simple bridge.
>
>     What have I missed?
>
>     Thanks in advance
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From ocset at the800group.com  Fri Apr 13 11:09:04 2012
From: ocset at the800group.com (ocset)
Date: Fri, 13 Apr 2012 15:09:04 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
Message-ID: <4F87D110.50103@the800group.com>

Hi

I have successfully managed to get FreeSwitch to bridge an incoming call 
on a PSTN line to an external mobile phone using a second PSTN line with 
the help of a GXW4104 FSO gateway.

I am now having a serious issue where the call never gets terminated 
when the parties hang up. The only way to end the call is to either 
shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a 
SIP gateway (PennyTel) instead of the second PSTN line, then it works as 
expected when both parties hang up.

Here is what I have tested

1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external 
mobile phone -- call end when both parties hang up.
2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external 
mobile phone -- call never end, even when both parties hang up.

Is there a parameter I should be setting to help solve this issue or 
some code that I should implement to test for this situation?

I have also noticed that with test scenario 1 (PSTN line + SIP Gateway), 
if the person being called does not end the call on their side, then 
that leg of the bridge remains active.

ps. The second PSTN line is not a real copper line. It is a PennyTel  
VOIP line that is managed by the modem and connected to the GXW4104 via 
phone cable. So, as far as the GXW4104 is concerned, it is just another 
POTS line. I don't have a second PSTN phone line so can't test any other 
way.

Thanks in advance
O.


From anita.hall at simmortel.com  Fri Apr 13 11:25:37 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Fri, 13 Apr 2012 12:55:37 +0530
Subject: [Freeswitch-users] T.30 Fax and Echo cancellation
Message-ID: <CAGKPtBu4Gpfp9VuMS4Cu73m_2n+XLfUCoVWL78gyWh=WjntXKQ@mail.gmail.com>

Hi

I know that all echo cancelers are supposed to shut down when they hear the
CNG or CED fax tone. Is this always a good thing ? I know that T.30 is
supposed to be half duplex, so echo should not be there but in my session
recordings, I have seen both sides trying to send at the same time (is that
a sign of some other bad thing?)

What if my T.30 fax is failing due to echo ?

Check out this article
http://connectedplanetonline.com/mag/telecom_oil_water_fax/


Thanks,
Anita.
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From Daniel.Knaggs at realitysolutions.co.uk  Fri Apr 13 11:35:30 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Fri, 13 Apr 2012 07:35:30 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <8C26A4FDAE599041A13EB499117D3C287CA27576@EX-MB-1.corp.atlasnetworks.us>
References: <8C26A4FDAE599041A13EB499117D3C287CA27576@EX-MB-1.corp.atlasnetworks.us>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05CF91@RSV16.realitysolutions.local>

That's very interesting, Nathan.

Now that makes three of us, two using the Digium cards and one using a Sangoma.


It very much looks like something FreeSwitch is doing, perhaps the DTMF detection is far too sensitive?  Are DTMF tones for an ISDN line inband or do they come in on the D-channel?


Michael - any ideas what we can all do?




From: Nathan Eisenberg [mailto:nathan at atlasnetworks.us]
Sent: 13 April 2012 00:21
To: Daniel Knaggs
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

I'm having the exact issue you're describing, with the same card.  However, what's interesting to me is that if I fire up Asterisk instead of freeswitch, the phantom DTMF problem disappears entirely.

So it's *something* inside Freeswitch, at least for me.  Would be interesting if you're able to do a similar test.
Nathan Eisenberg
Atlas Networks | Sr. Systems Administrator
office: 206.577.3078  |  www.atlasnetworks.us
 D


Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel:    01482 373104
Mobile: 07932 408313
Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
http://





From gcd at i.ph  Fri Apr 13 11:40:25 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Fri, 13 Apr 2012 15:40:25 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <4F87D110.50103@the800group.com>
References: <4F87D110.50103@the800group.com>
Message-ID: <CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>

check the line supervision signalling of your GSM and the GXW4104. there
are several disconnect signalling used: current disconnect, polarity
reversal or busy tones. your FXO port must be set accordingly.

On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com> wrote:

> Hi
>
> I have successfully managed to get FreeSwitch to bridge an incoming call
> on a PSTN line to an external mobile phone using a second PSTN line with
> the help of a GXW4104 FSO gateway.
>
> I am now having a serious issue where the call never gets terminated
> when the parties hang up. The only way to end the call is to either
> shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a
> SIP gateway (PennyTel) instead of the second PSTN line, then it works as
> expected when both parties hang up.
>
> Here is what I have tested
>
> 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external
> mobile phone -- call end when both parties hang up.
> 2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external
> mobile phone -- call never end, even when both parties hang up.
>
> Is there a parameter I should be setting to help solve this issue or
> some code that I should implement to test for this situation?
>
> I have also noticed that with test scenario 1 (PSTN line + SIP Gateway),
> if the person being called does not end the call on their side, then
> that leg of the bridge remains active.
>
> ps. The second PSTN line is not a real copper line. It is a PennyTel
> VOIP line that is managed by the modem and connected to the GXW4104 via
> phone cable. So, as far as the GXW4104 is concerned, it is just another
> POTS line. I don't have a second PSTN phone line so can't test any other
> way.
>
> Thanks in advance
> O.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vitaliy.davudov at vts24.ru  Fri Apr 13 13:18:24 2012
From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=)
Date: Fri, 13 Apr 2012 13:18:24 +0400
Subject: [Freeswitch-users] Variable continue on fail
Message-ID: <4F87EF60.3070105@vts24.ru>

Hi all!

In my dialplan I've included variable continue on fail:

<extension name="Dial Out SIP-M200 Mobile numbers">
<condition field="destination_number" expression="^(89\d{9})$">
<action application="export" data="t38_passthru=true"/>
<action application="set" 
data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
<action application="info"/>
<action application="bridge" data="sofia/gateway/SIP_M200/$1"/>
<action application="limit_execute" data="hash limit SMG-1016M 60 bridge 
sofia/gateway/SMG-1016M/$1"/>
</condition>
</extension>

And if FS recieve from first gateway USER_BUSY, then FS  try to bridge 
this call to another gateway. Although in line <action application="set" 
data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> 
there is no code Q.850 = 17.
How resolve this issue?

-- 
Best regards,
Vitaly Davudov
"VIP-TELECOM-SERVICE" Ltd.
("ETERIA" Group of companies)
http://www.vts24.ru




From vfclists at gmail.com  Fri Apr 13 13:38:38 2012
From: vfclists at gmail.com (Frank Church)
Date: Fri, 13 Apr 2012 10:38:38 +0100
Subject: [Freeswitch-users] what is the purpose and value of the files in
	the db directory?
Message-ID: <CAE3mkNDJie83wpu3r9xOmVDGvcBnvUDUgsubkBPhtNuMghASZg@mail.gmail.com>

I have been getting an occasional problem when freeswitch fails to
start correctly. I googled around and found out that its was caused by
the corruption of files in the db directory.

What runtime related information is stored in those files? Are they
useful for some logging or analytical value?

Can an alternate database system be used for them?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From anton.jugatsu at gmail.com  Fri Apr 13 13:44:13 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 13 Apr 2012 13:44:13 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <4F87EF60.3070105@vts24.ru>
References: <4F87EF60.3070105@vts24.ru>
Message-ID: <CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>

Ok, i got it. Even that there is no USER_BUSY at continue_on_fail variable,
FS still tries to reach the second action, am i right?

So, for better debugging, i suggest to paste full call log with enabled
siptrace and /log 7 to pastebin.freeswitch.org.

13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? <
vitaliy.davudov at vts24.ru> ???????:

> Hi all!
>
> In my dialplan I've included variable continue on fail:
>
> <extension name="Dial Out SIP-M200 Mobile numbers">
> <condition field="destination_number" expression="^(89\d{9})$">
> <action application="export" data="t38_passthru=true"/>
> <action application="set"
> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
> <action application="info"/>
> <action application="bridge" data="sofia/gateway/SIP_M200/$1"/>
> <action application="limit_execute" data="hash limit SMG-1016M 60 bridge
> sofia/gateway/SMG-1016M/$1"/>
> </condition>
> </extension>
>
> And if FS recieve from first gateway USER_BUSY, then FS  try to bridge
> this call to another gateway. Although in line <action application="set"
> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
> there is no code Q.850 = 17.
> How resolve this issue?
>
> --
> Best regards,
> Vitaly Davudov
> "VIP-TELECOM-SERVICE" Ltd.
> ("ETERIA" Group of companies)
> http://www.vts24.ru
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vfclists at gmail.com  Fri Apr 13 14:01:11 2012
From: vfclists at gmail.com (Frank Church)
Date: Fri, 13 Apr 2012 11:01:11 +0100
Subject: [Freeswitch-users] Need some help with dial string matching?
Message-ID: <CAE3mkNA9GtmK4y8vLh+=+D=E5Ly4CY61qDphjsxCKDNEZHT-pg@mail.gmail.com>

I set up a basic installation of freeswitch, but because I want to learn
more I haven't removed the existing extensions, but I created a context to
send all numbers beginnng with 0 through an external provider.

<include>
>     <!--   $CONFIGDIR/dialplan/provider.xml
>
> -->
> <context name="provider">
>    <extension name="outbound">
>      <condition field="destination_number" expression="^0?(\d+)$">
>        <action application="bridge" data="sofia/gateway/provider/0$1"/>
>      </condition>
>    </extension>
>  </context>
> </include>
>

I noticed that numbers without a leading zero are matched and dialled.
Anyway I think I realize that the expression should start wih '^0+' instead
of ^0?'. How then do I dial the whole string instead of using 0$1? Because
that is what is adding the leading 0 to the numbers without it.

I also want to ensure that a minimal number of digits are dialled before
the call is allowed. Do I need to add them literally like ^0+dddddd or is
there a shortcut for specifying a minimal number of digits?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From vfclists at gmail.com  Fri Apr 13 14:18:10 2012
From: vfclists at gmail.com (Frank Church)
Date: Fri, 13 Apr 2012 11:18:10 +0100
Subject: [Freeswitch-users] Need some help with dial string matching?
In-Reply-To: <CAE3mkNA9GtmK4y8vLh+=+D=E5Ly4CY61qDphjsxCKDNEZHT-pg@mail.gmail.com>
References: <CAE3mkNA9GtmK4y8vLh+=+D=E5Ly4CY61qDphjsxCKDNEZHT-pg@mail.gmail.com>
Message-ID: <CAE3mkNDMw7-3NQnjeXJq=04G9WLntSKiPc6WfK4fhmj2N+jVcQ@mail.gmail.com>

On 13 April 2012 11:01, Frank Church <vfclists at gmail.com> wrote:
>
> I set up a basic installation of freeswitch, but because I want to learn
> more I haven't removed the existing extensions, but I created a context to
> send all numbers beginnng with 0 through an external provider.
>
>> <include>
>> ? ? <!-- ? $CONFIGDIR/dialplan/provider.xml
>>
>> -->
>> <context name="provider">
>> ? ?<extension name="outbound">
>> ? ? ?<condition field="destination_number" expression="^0?(\d+)$">
>> ? ? ? ?<action application="bridge" data="sofia/gateway/provider/0$1"/>
>> ? ? ?</condition>
>> ? ?</extension>
>> ?</context>
>> </include>
>
>
> I noticed that numbers without a leading zero are matched and dialled.
> Anyway I think I realize that the expression should start wih '^0+' instead
> of ^0?'. How then do I dial the whole string instead of using 0$1? Because
> that is what is adding the leading 0 to the numbers without it.
>
> I also want to ensure that a minimal number of digits are dialled before the
> call is allowed. Do I need to add them literally like ^0+dddddd or is there
> a shortcut for specifying a minimal number of digits?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>

http://wiki.freeswitch.org/wiki/Regular_Expression has the info I need.

Thanks

-- 
Frank Church

=======================
http://devblog.brahmancreations.com


From herman.griffin at gmail.com  Fri Apr 13 15:24:26 2012
From: herman.griffin at gmail.com (Herman Griffin)
Date: Fri, 13 Apr 2012 04:24:26 -0700
Subject: [Freeswitch-users] Doing basic arithmetic in the xml dialplan
	without a language
Message-ID: <CAGTy0Gyw-H0VdJAZ5wa5g9zbKHqV=fBG=HHD4iY_n6yOU0bbpA@mail.gmail.com>

Is there a way to do basic arithmetic within the dialplan without running a
mod_language script. For instance, to increment a counter on every call. Or
to use the modula or division.

Herman
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From vipkilla at gmail.com  Fri Apr 13 15:52:34 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 13 Apr 2012 07:52:34 -0400
Subject: [Freeswitch-users] Doing basic arithmetic in the xml dialplan
 without a language
In-Reply-To: <CAGTy0Gyw-H0VdJAZ5wa5g9zbKHqV=fBG=HHD4iY_n6yOU0bbpA@mail.gmail.com>
References: <CAGTy0Gyw-H0VdJAZ5wa5g9zbKHqV=fBG=HHD4iY_n6yOU0bbpA@mail.gmail.com>
Message-ID: <CAC-LwPP0arzUH-ayqESMKvOwp=nrvMZFmvuLjaqHf-rz2ri8ng@mail.gmail.com>

This is an example of adding 1 to the 'inc' variable. If 'inc' is not
defined, it will become 1. If it is 1, it will become 2, etc...
<action application="set" data="inc=${expr(${inc}+1)}"/>

On Fri, Apr 13, 2012 at 7:24 AM, Herman Griffin
<herman.griffin at gmail.com> wrote:
> Is there a way to do basic arithmetic within the dialplan without running a
> mod_language script. For instance, to increment a counter on every call. Or
> to use the modula or division.
>
> Herman
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From hkalyoncu at gmail.com  Fri Apr 13 16:37:25 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Fri, 13 Apr 2012 15:37:25 +0300
Subject: [Freeswitch-users] dynamic dialplan alternatives
Message-ID: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>

we are currently have two fs boxes load balanced by osips.
in each fs we have dynamic dialplan with mod_xml_curl using php & apache &
mysql.
this conf is doing just fine with current load (about 20 cps) but with the
increasing of incoming traffic, it shows some performance issues.
we want to increase our call throughput.

i searched through fs site and mailing list and came up with following
options:

1) using lua or(python?) to serve dialplan instead of mod_xml_curl
2) writing a dialplan module(something like mod_xml_curl) in c which will
do all
the db lookups etc..
3) using mod_erlang_event in outbound mode & spawning several erlang
workers to do db lookups etc..
4) using mod_event_socket in outbound mode & making db lookups on a
different server.

there is little information about 3 and 4 (only relative wiki pages)
so im not sure even they are make any sense.

before choosing one of above options i want to ask that
which one of these would you prefer to increase performance. or do you
have another suggestion other than these?


thanks & best regards,
H?seyin Kalyoncu
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From vipkilla at gmail.com  Fri Apr 13 16:46:20 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 13 Apr 2012 08:46:20 -0400
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
Message-ID: <CAC-LwPPApCTWya0LNOOxdbSyqOVPCsRHdx=vzL5ABsff6w6nCA@mail.gmail.com>

> but with the increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.
>

What sort of performance issues? High CPU load? Delayed calls?


From vitaliy.davudov at vts24.ru  Fri Apr 13 16:57:09 2012
From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=)
Date: Fri, 13 Apr 2012 16:57:09 +0400
Subject: [Freeswitch-users] Variable continue on fail
In-Reply-To: <CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
References: <4F87EF60.3070105@vts24.ru>
	<CANBhyPAeQxnObEWS1YoD_u6j9D-wB_pZWGL+WHp9bhDt_QO_5g@mail.gmail.com>
Message-ID: <4F8822A5.9090402@vts24.ru>

Yes, you are right!

I did it: http://pastebin.freeswitch.org/18863

Additionally:
I've included in this extension new line:

<extension name="Dial Out SIP-M200 Mobile numbers">
<condition field="destination_number" expression="^(89\d{9})$">
<action application="export" data="t38_passthru=true"/>
*<action application="set" data="hangup_after_bridge=true"/>*
<action application="set" 
data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
<action application="info"/>
<action application="bridge" data="sofia/gateway/SIP_M200/$1"/>
<action application="limit_execute" data="hash limit SMG-1016M 60 bridge 
sofia/gateway/SMG-1016M/$1"/>
</condition>
</extension>

Without that line a similar situation occurs if FS recieve 
/NORMAL_CLEARING./

13.04.2012 13:44, Anton Kvashenkin ???????:
> Ok, i got it. Even that there is no USER_BUSY at continue_on_fail 
> variable, FS still tries to reach the second action, am i right?
>
> So, for better debugging, i suggest to paste full call log with 
> enabled siptrace and /log 7 to pastebin.freeswitch.org 
> <http://pastebin.freeswitch.org>.
>
> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? 
> <vitaliy.davudov at vts24.ru <mailto:vitaliy.davudov at vts24.ru>> ???????:
>
>     Hi all!
>
>     In my dialplan I've included variable continue on fail:
>
>     <extension name="Dial Out SIP-M200 Mobile numbers">
>     <condition field="destination_number" expression="^(89\d{9})$">
>     <action application="export" data="t38_passthru=true"/>
>     <action application="set"
>     data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
>     <action application="info"/>
>     <action application="bridge" data="sofia/gateway/SIP_M200/$1"/>
>     <action application="limit_execute" data="hash limit SMG-1016M 60
>     bridge
>     sofia/gateway/SMG-1016M/$1"/>
>     </condition>
>     </extension>
>
>     And if FS recieve from first gateway USER_BUSY, then FS  try to bridge
>     this call to another gateway. Although in line <action
>     application="set"
>     data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/>
>     there is no code Q.850 = 17.
>     How resolve this issue?
>
>     --
>     Best regards,
>     Vitaly Davudov
>     "VIP-TELECOM-SERVICE" Ltd.
>     ("ETERIA" Group of companies)
>     http://www.vts24.ru
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
? ?????????? ???????????,
??????? ??????? ?????????
??? "???-???????-??????"
(?????? ???????? "ETERIA")
http://www.vts24.ru
???: (495) 989-47-00

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From lazyvirus at gmx.com  Fri Apr 13 17:12:16 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Fri, 13 Apr 2012 15:12:16 +0200
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
Message-ID: <20120413151216.6b3be579@anubis.defcon1>

On Fri, 13 Apr 2012 15:37:25 +0300
huseyin kalyoncu <hkalyoncu at gmail.com> wrote:

Hi Huseyin,

first, I'm not a FS specialist, so don't think my words are the
solution, only what I would do.

> we are currently have two fs boxes load balanced by osips.
> in each fs we have dynamic dialplan with mod_xml_curl using php & apache &
> mysql.
> this conf is doing just fine with current load (about 20 cps) but with the
> increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.

I'd jettison apache and replace it with i.e. nginx or lighttpd
(memory and latency matter, see the many tests on the web).

IF DB isn't very large, I'd also switch it to sqlite3 and copy it
to/from memory for (much) faster access (just in case, sqlite3
support foreign keys since V.3.6.19).

I'd also "cut" the dialplan between simple parts (on disk) and
subject to changes parts (on DB).

But first of all, I'd try to time all operations to discover where
the bottle(s) neck(s) is/are and make sure I've got enough RAM (not
for FS, but for the whole shebang around it).

My 2 ? (not on the head!)

Jean-Yves
-- 
A male gynecologist is like an auto mechanic who has never owned a
car. -- Carrie Snow


From lists at telefaks.de  Fri Apr 13 17:48:59 2012
From: lists at telefaks.de (Peter Steinbach)
Date: Fri, 13 Apr 2012 15:48:59 +0200
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
Message-ID: <4F882ECB.3070108@telefaks.de>

I would also consider which language to use. Scripting php may be slow 
in Apache. We have had good performance dates with Rails. With some 
caching we achieved up to ~500 requests/sec using Apache load balancer 
and mongel_cluster.

Best regards
Peter


Am 13.04.2012 14:37, schrieb huseyin kalyoncu:
> we are currently have two fs boxes load balanced by osips.
> in each fs we have dynamic dialplan with mod_xml_curl using php & 
> apache & mysql.
> this conf is doing just fine with current load (about 20 cps) but with the
> increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.
>
> i searched through fs site and mailing list and came up with following 
> options:
>
> 1) using lua or(python?) to serve dialplan instead of mod_xml_curl
> 2) writing a dialplan module(something like mod_xml_curl) in c which 
> will do all
> the db lookups etc..
> 3) using mod_erlang_event in outbound mode & spawning several erlang
> workers to do db lookups etc..
> 4) using mod_event_socket in outbound mode & making db lookups on a
> different server.
>
> there is little information about 3 and 4 (only relative wiki pages)
> so im not sure even they are make any sense.
>
> before choosing one of above options i want to ask that
> which one of these would you prefer to increase performance. or do you
> have another suggestion other than these?
>
>
> thanks & best regards,
> H?seyin Kalyoncu
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


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From sunwood360 at gmail.com  Fri Apr 13 18:17:59 2012
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Fri, 13 Apr 2012 07:17:59 -0700
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
Message-ID: <CAKq5mS5HbeAkYiiXmtOhnVLaShjrv1ag=R=br_YAwaiUbNve3g@mail.gmail.com>

Choice 3 or 4 should be your long term strategic direction.

On Apr 13, 2012 8:39 AM, "huseyin kalyoncu" <hkalyoncu at gmail.com> wrote:
>
> we are currently have two fs boxes load balanced by osips.
> in each fs we have dynamic dialplan with mod_xml_curl using php & apache
& mysql.
> this conf is doing just fine with current load (about 20 cps) but with the
> increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.
>
> i searched through fs site and mailing list and came up with following
options:
>
> 1) using lua or(python?) to serve dialplan instead of mod_xml_curl
> 2) writing a dialplan module(something like mod_xml_curl) in c which will
do all
> the db lookups etc..
> 3) using mod_erlang_event in outbound mode & spawning several erlang
> workers to do db lookups etc..
> 4) using mod_event_socket in outbound mode & making db lookups on a
> different server.
>
> there is little information about 3 and 4 (only relative wiki pages)
> so im not sure even they are make any sense.
>
> before choosing one of above options i want to ask that
> which one of these would you prefer to increase performance. or do you
> have another suggestion other than these?
>
>
> thanks & best regards,
> H?seyin Kalyoncu
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From krice at freeswitch.org  Fri Apr 13 20:16:59 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 13 Apr 2012 11:16:59 -0500
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CAE3mkNDJie83wpu3r9xOmVDGvcBnvUDUgsubkBPhtNuMghASZg@mail.gmail.com>
Message-ID: <CBADBBAB.311D1%krice@freeswitch.org>

The files in the db directory are used for a variety of things...
Most of the files are for storing runtime information that can be nuked
across reboots the exception to that is the voicemail db... If you nuke it
you lose all your voicemail metadata effectively resetting all the voicemail
boxes.

Data stored in there is used for various things like show calls and show
channels so that we don't have to lock memory structures just to show you
what calls are active and what state they are in. (this was a lessoned
learned from years of seeing "avoiding deadlock" from a simple show me the
calls.

As far as using an alternate database yes you can... See the wiki for core
odbc docs. This applies not just to the core but to several other modules
individually. 

Also as a side note, if you are not using the voicemail module, you can
mount the db directory on a ram drive. This has been shown to increse
performance and reduce disk IO a fair bit...



On 4/13/12 4:38 AM, "Frank Church" <vfclists at gmail.com> wrote:

> I have been getting an occasional problem when freeswitch fails to
> start correctly. I googled around and found out that its was caused by
> the corruption of files in the db directory.
> 
> What runtime related information is stored in those files? Are they
> useful for some logging or analytical value?
> 
> Can an alternate database system be used for them?




From Hector.Geraldino at ipsoft.com  Fri Apr 13 20:49:14 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Fri, 13 Apr 2012 12:49:14 -0400
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAKq5mS5HbeAkYiiXmtOhnVLaShjrv1ag=R=br_YAwaiUbNve3g@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAKq5mS5HbeAkYiiXmtOhnVLaShjrv1ag=R=br_YAwaiUbNve3g@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022E872E93@NY1-EXMB-01.ip-soft.net>

I would go with option 4 + a good caching strategy on the socket application side. Having an specialized application without the HTTP overhead should be faster (at least in theory). And if you can avoid the db lookup by having a second level cache integrated on the app, be prepared to near real-time responses :)

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of envelopes envelopes
Sent: Friday, April 13, 2012 10:18 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] dynamic dialplan alternatives


Choice 3 or 4 should be your long term strategic direction.

On Apr 13, 2012 8:39 AM, "huseyin kalyoncu" <hkalyoncu at gmail.com<mailto:hkalyoncu at gmail.com>> wrote:
>
> we are currently have two fs boxes load balanced by osips.
> in each fs we have dynamic dialplan with mod_xml_curl using php & apache & mysql.
> this conf is doing just fine with current load (about 20 cps) but with the
> increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.
>
> i searched through fs site and mailing list and came up with following options:
>
> 1) using lua or(python?) to serve dialplan instead of mod_xml_curl
> 2) writing a dialplan module(something like mod_xml_curl) in c which will do all
> the db lookups etc..
> 3) using mod_erlang_event in outbound mode & spawning several erlang
> workers to do db lookups etc..
> 4) using mod_event_socket in outbound mode & making db lookups on a
> different server.
>
> there is little information about 3 and 4 (only relative wiki pages)
> so im not sure even they are make any sense.
>
> before choosing one of above options i want to ask that
> which one of these would you prefer to increase performance. or do you
> have another suggestion other than these?
>
>
> thanks & best regards,
> H?seyin Kalyoncu
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at 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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From lloyd.aloysius at gmail.com  Fri Apr 13 21:17:21 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Fri, 13 Apr 2012 13:17:21 -0400
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022E872E93@NY1-EXMB-01.ip-soft.net>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAKq5mS5HbeAkYiiXmtOhnVLaShjrv1ag=R=br_YAwaiUbNve3g@mail.gmail.com>
	<6A6B4C284AD15042B429EB9D904544AD022E872E93@NY1-EXMB-01.ip-soft.net>
Message-ID: <CAGvuWYYt0CGDvzBda9-90nAW6+81wHqncmUhs4YdO_-c8WCaTg@mail.gmail.com>

I have the same question as Vik. What kind of performance issue? What is
the hardware Configuration Memory ,CPU?

Lloyd
*
*



On Fri, Apr 13, 2012 at 12:49 PM, Hector Geraldino <
Hector.Geraldino at ipsoft.com> wrote:

> I would go with option 4 + a good caching strategy on the socket
> application side. Having an specialized application without the HTTP
> overhead should be faster (at least in theory). And if you can avoid the db
> lookup by having a second level cache integrated on the app, be prepared to
> near real-time responses J****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *envelopes
> envelopes
> *Sent:* Friday, April 13, 2012 10:18 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] dynamic dialplan alternatives****
>
> ** **
>
> Choice 3 or 4 should be your long term strategic direction.****
>
> On Apr 13, 2012 8:39 AM, "huseyin kalyoncu" <hkalyoncu at gmail.com> wrote:
> >
> > we are currently have two fs boxes load balanced by osips.
> > in each fs we have dynamic dialplan with mod_xml_curl using php & apache
> & mysql.
> > this conf is doing just fine with current load (about 20 cps) but with
> the
> > increasing of incoming traffic, it shows some performance issues.
> > we want to increase our call throughput.
> >
> > i searched through fs site and mailing list and came up with following
> options:
> >
> > 1) using lua or(python?) to serve dialplan instead of mod_xml_curl
> > 2) writing a dialplan module(something like mod_xml_curl) in c which
> will do all
> > the db lookups etc..
> > 3) using mod_erlang_event in outbound mode & spawning several erlang
> > workers to do db lookups etc..
> > 4) using mod_event_socket in outbound mode & making db lookups on a
> > different server.
> >
> > there is little information about 3 and 4 (only relative wiki pages)
> > so im not sure even they are make any sense.
> >
> > before choosing one of above options i want to ask that
> > which one of these would you prefer to increase performance. or do you
> > have another suggestion other than these?
> >
> >
> > thanks & best regards,
> > H?seyin Kalyoncu
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gabe at gundy.org  Fri Apr 13 21:22:23 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 Apr 2012 11:22:23 -0600
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
Message-ID: <CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>

On Fri, Apr 13, 2012 at 6:37 AM, huseyin kalyoncu <hkalyoncu at gmail.com> wrote:
> we are currently have two fs boxes load balanced by osips.
> in each fs we have dynamic dialplan with mod_xml_curl using php & apache &
> mysql. this conf is doing just fine with current load (about 20 cps) but with the
> increasing of incoming traffic, it shows some performance issues.
> we want to increase our call throughput.

You don't really give us enough to go on. A few questions that I have
when I read this...

* Are the DB and HTTP servers running on stand alone servers, or are
they running on the FS server?
* What kind of hardware are they each running on? RAM, CPU, disks?
* What kind of load do the servers have when you see performance issues?
* What have you done to optimize your web stack (other HTTP servers
etc)? Are you caching where you can?
* Have you analyzed your SQL to find any slow queries?

Now, I don't expect you to answer all of these in this email thread,
but I'll bet if you reviewed them, you'd be able to bump your
performance without having to revisit your current approach.


> i searched through fs site and mailing list and came up with following
> options:

I'll comment on this generically, but I can't really say what each
approach will (or will not) do for your current situation.


> 1) using lua or (python?) to serve dialplan instead of mod_xml_curl

This works. Lua is lighter so it may give you better performance than
Python (this is a non-issue if you're using Python in a long-living
process with event socket or mod_xml_curl). Regardless of languages,
this approach has a drawback in that processing is done on the same
server that's running FreeSWITCH.


> 2) writing a dialplan module (something like mod_xml_curl) in c which will do
> all the db lookups etc..

That seems a little extreme. Many people have scaled way up without
resorting to that approach.


> 3) using mod_erlang_event in outbound mode & spawning several erlang
> workers to do db lookups etc..

This is about the same as #4 (give or take some Erlang magic).


> 4) using mod_event_socket in outbound mode & making db lookups on a
> different server.

This is a fine approach when controlling the call, but you still need
some basic dialplan to get the calls going where they need to go. I
don't see it as the right way to solve your scaling problem, but it
might be part of the overall solution.


It sounds like you might be giving up on mod_xml_curl too soon.
Scaling HTTP is a well known problem with lots of tools available to
help you. When developing for mod_xml_curl, try returning the simplest
bit of dialplan you can and do all the work (logic, DB, etc.) on the
HTTP server. And don't forget that in *any* scaling scenario, you will
need to bring hardware, *real* hardware :)

Good luck and let us know how you end up solving the problem.


Best,
Gabe


From gabe at gundy.org  Fri Apr 13 21:23:40 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Fri, 13 Apr 2012 11:23:40 -0600
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CBADBBAB.311D1%krice@freeswitch.org>
References: <CAE3mkNDJie83wpu3r9xOmVDGvcBnvUDUgsubkBPhtNuMghASZg@mail.gmail.com>
	<CBADBBAB.311D1%krice@freeswitch.org>
Message-ID: <CAMwZ-tJZoXOHuemkHYbBAvihBKwuAUjOjB0PHAx5qzdUGPm-gg@mail.gmail.com>

On Fri, Apr 13, 2012 at 10:16 AM, Ken Rice <krice at freeswitch.org> wrote:
> Data stored in there is used for various things like show calls and show
> channels so that we don't have to lock memory structures just to show you
> what calls are active and what state they are in. (this was a lessoned
> learned from years of seeing "avoiding deadlock" from a simple show me the
> calls.

I don't want to get all nostalgic, but I miss those days :)


Gabe


From ifoundthetao at gmail.com  Fri Apr 13 21:36:47 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Fri, 13 Apr 2012 12:36:47 -0500
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
Message-ID: <4F88642F.9010706@gmail.com>

Hey all,

I know that there is a good amount of information out there on how to do 
faxing via FreeSWITCH, but I'm still finding myself a bit confused.

I'm currently trying to figure out the process by which I would need to 
do this.  I'll detail my setup, and ask questions.  I've read 
documentation, but I'm still missing something.

Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM, 
and I'm the only user.  I'm using FlowRoute, and I have a couple of 
unused numbers.  Pretty simple, nothing out of the ordinary (except 
maybe the VM part?).

At some point in the next six months or so, I'll be getting a couple of 
dedicated servers here for various things, and I'll begin moving my VM 
servers from my main computer to those, as actual hardware hosts.

So my fax machine is built into an "All-in-one" Network 
Printer/Scanner/Fax (Brother MFC-7360N).

Here's my main question, do I need to get a special card to run a cable 
from my computer to the Fax?

Is it better to get a dedicated fax machine to use, or are there 
specialized fax machines for Fax over IP?

Thanks guys and gals,

Tim

-- 
'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed



From bdfoster at endigotech.com  Fri Apr 13 21:51:55 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 13 Apr 2012 13:51:55 -0400
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F88642F.9010706@gmail.com>
References: <4F88642F.9010706@gmail.com>
Message-ID: <CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>

I've looked at that model, and it doesn't actually say this but I'm
assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
my faxing using SpanDSP, but basically what you would need is an ATA that
supports T.38, as well as a provider that supports T.38. Flowroute seems to
be pretty reliable (that's who I use). I have a fax machine that I've
played around with here, and I'm using a Grandstream HT286 ATA to connect
it to FreeSWITCH.

Having said all of that, it's highly dependent on your setup. Questions:

1. Are you going to have dedicated fax numbers?
1.1. If so, will each fax machine have it's own fax number?
2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?

If there's any other details you feel are related, please mention those
too. Realize that FS can do faxing on it's own without the need for a fax
machine. Most, however, decide not to go down that route mostly because the
fax machine is an interface most are comfortable with. You can even do
outbound faxes on a fax machine, and get FS to handle incoming faxes and
email them, store them, and even print them to a network printer.

We're more than happy to help!

-BDF

On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:

> Hey all,
>
> I know that there is a good amount of information out there on how to do
> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>
> I'm currently trying to figure out the process by which I would need to
> do this.  I'll detail my setup, and ask questions.  I've read
> documentation, but I'm still missing something.
>
> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
> and I'm the only user.  I'm using FlowRoute, and I have a couple of
> unused numbers.  Pretty simple, nothing out of the ordinary (except
> maybe the VM part?).
>
> At some point in the next six months or so, I'll be getting a couple of
> dedicated servers here for various things, and I'll begin moving my VM
> servers from my main computer to those, as actual hardware hosts.
>
> So my fax machine is built into an "All-in-one" Network
> Printer/Scanner/Fax (Brother MFC-7360N).
>
> Here's my main question, do I need to get a special card to run a cable
> from my computer to the Fax?
>
> Is it better to get a dedicated fax machine to use, or are there
> specialized fax machines for Fax over IP?
>
> Thanks guys and gals,
>
> Tim
>
> --
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From ifoundthetao at gmail.com  Fri Apr 13 22:09:04 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Fri, 13 Apr 2012 13:09:04 -0500
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
Message-ID: <4F886BC0.2010306@gmail.com>

You've raised some good questions, and good points.

This model does have an RJ-11, and I believe you are correct.  I don't 
have an ATA yet, but I'll get the ATA model that you have (Grandstream 
HT286 ATA).

1 - I'll do a dedicated fax number
1.1 - Each fax machine will have its own number
2 -  As it stands, I'm going to be running this over VoIP

As far as faxing via FS w/o the machine goes, I wasn't really aware of 
that.  I think I knew it, but it didn't click.  After I get the ATA 
setup working, I think I'll set up another version where FS is handling 
it w/o a machine.  That way I'll have both types of experience under my 
belt.

Actually, as I think about it more.. it probably makes more sense to set 
up the environment where the machine isn't present first (as I don't 
have the ATA on hand now)...

Thanks!

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 4/13/2012 12:51 PM, Brian Foster wrote:
> I've looked at that model, and it doesn't actually say this but I'm 
> assuming you connect an RJ-11 to the fax modem inside the all-in-one. 
> I do my faxing using SpanDSP, but basically what you would need is an 
> ATA that supports T.38, as well as a provider that supports T.38. 
> Flowroute seems to be pretty reliable (that's who I use). I have a fax 
> machine that I've played around with here, and I'm using a Grandstream 
> HT286 ATA to connect it to FreeSWITCH.
>
> Having said all of that, it's highly dependent on your setup. Questions:
>
> 1. Are you going to have dedicated fax numbers?
> 1.1. If so, will each fax machine have it's own fax number?
> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>
> If there's any other details you feel are related, please mention 
> those too. Realize that FS can do faxing on it's own without the need 
> for a fax machine. Most, however, decide not to go down that route 
> mostly because the fax machine is an interface most are comfortable 
> with. You can even do outbound faxes on a fax machine, and get FS to 
> handle incoming faxes and email them, store them, and even print them 
> to a network printer.
>
> We're more than happy to help!
>
> -BDF
>
> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton 
> <ifoundthetao at gmail.com <mailto:ifoundthetao at gmail.com>> wrote:
>
>     Hey all,
>
>     I know that there is a good amount of information out there on how
>     to do
>     faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>
>     I'm currently trying to figure out the process by which I would
>     need to
>     do this.  I'll detail my setup, and ask questions.  I've read
>     documentation, but I'm still missing something.
>
>     Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of
>     RAM,
>     and I'm the only user.  I'm using FlowRoute, and I have a couple of
>     unused numbers.  Pretty simple, nothing out of the ordinary (except
>     maybe the VM part?).
>
>     At some point in the next six months or so, I'll be getting a
>     couple of
>     dedicated servers here for various things, and I'll begin moving my VM
>     servers from my main computer to those, as actual hardware hosts.
>
>     So my fax machine is built into an "All-in-one" Network
>     Printer/Scanner/Fax (Brother MFC-7360N).
>
>     Here's my main question, do I need to get a special card to run a
>     cable
>     from my computer to the Fax?
>
>     Is it better to get a dedicated fax machine to use, or are there
>     specialized fax machines for Fax over IP?
>
>     Thanks guys and gals,
>
>     Tim
>
>     --
>     'We who cut mere stones must always be envisioning cathedrals.'
>     Quarry Worker's Creed
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for 
> those listed in the "To:", "CC:", and/or "BCC:" fields of the message 
> header. If you are not the intended recipient you are notified that 
> disclosing, copying, distributing or taking any action in reliance on 
> the contents of this information is strictly prohibited. E-mail 
> transmission cannot be guaranteed to be secure or error-free as 
> information could be intercepted, corrupted, lost, destroyed, arrive 
> late or incomplete, or contain viruses. The sender therefore does not 
> accept liability for any errors or omissions in the contents of this 
> message, which arise as a result of e-mail transmission. If 
> verification is required please request a hard-copy version.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Fri Apr 13 22:29:40 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 13 Apr 2012 14:29:40 -0400
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F886BC0.2010306@gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
Message-ID: <CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>

Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent on
network conditions and carrier support and implementation.

With that out of the way, the best place to start doing fax using your FS
machine (only) would be here:

Place this in your public context (maybe it shouldn't stay there, but this
is an extremely easy way to mess around with faxing):

<extension name="fax_receive">
  <condition field="context" expression="public"/>
  <condition field="destination_number" expression="^9978$">
    <action application="answer"/>
    <action application="playback" data="silence_stream://2000"/>
    <action application="set" data="fax_enable_t38_request=true"/>
    <action application="set" data="fax_enable_t38=true"/>
    <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
    <action application="hangup"/>
  </condition>
</extension>

If you want to run a script after the fax is recieved, put this line in
before the "rxfax":

    <action application="set" data="api_hangup_hook=system <scriptname>
<arg1> <arg2> <argN>"/>

Start with receiving faxes, and go from there. Sending faxes requires some
extra work, but not much. For more information about faxing, please visit
http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax

Obviously if you have any questions, please do not hesitate to ask!

-BDF

On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:

>  You've raised some good questions, and good points.
>
> This model does have an RJ-11, and I believe you are correct.  I don't
> have an ATA yet, but I'll get the ATA model that you have (Grandstream
> HT286 ATA).
>
> 1 - I'll do a dedicated fax number
> 1.1 - Each fax machine will have its own number
> 2 -  As it stands, I'm going to be running this over VoIP
>
> As far as faxing via FS w/o the machine goes, I wasn't really aware of
> that.  I think I knew it, but it didn't click.  After I get the ATA setup
> working, I think I'll set up another version where FS is handling it w/o a
> machine.  That way I'll have both types of experience under my belt.
>
> Actually, as I think about it more.. it probably makes more sense to set
> up the environment where the machine isn't present first (as I don't have
> the ATA on hand now)...
>
> Thanks!
>
>
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> On 4/13/2012 12:51 PM, Brian Foster wrote:
>
> I've looked at that model, and it doesn't actually say this but I'm
> assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
> my faxing using SpanDSP, but basically what you would need is an ATA that
> supports T.38, as well as a provider that supports T.38. Flowroute seems to
> be pretty reliable (that's who I use). I have a fax machine that I've
> played around with here, and I'm using a Grandstream HT286 ATA to connect
> it to FreeSWITCH.
>
> Having said all of that, it's highly dependent on your setup. Questions:
>
> 1. Are you going to have dedicated fax numbers?
> 1.1. If so, will each fax machine have it's own fax number?
> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>
> If there's any other details you feel are related, please mention those
> too. Realize that FS can do faxing on it's own without the need for a fax
> machine. Most, however, decide not to go down that route mostly because the
> fax machine is an interface most are comfortable with. You can even do
> outbound faxes on a fax machine, and get FS to handle incoming faxes and
> email them, store them, and even print them to a network printer.
>
> We're more than happy to help!
>
> -BDF
>
> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>
>> Hey all,
>>
>> I know that there is a good amount of information out there on how to do
>> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>>
>> I'm currently trying to figure out the process by which I would need to
>> do this.  I'll detail my setup, and ask questions.  I've read
>> documentation, but I'm still missing something.
>>
>> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
>> and I'm the only user.  I'm using FlowRoute, and I have a couple of
>> unused numbers.  Pretty simple, nothing out of the ordinary (except
>> maybe the VM part?).
>>
>> At some point in the next six months or so, I'll be getting a couple of
>> dedicated servers here for various things, and I'll begin moving my VM
>> servers from my main computer to those, as actual hardware hosts.
>>
>> So my fax machine is built into an "All-in-one" Network
>> Printer/Scanner/Fax (Brother MFC-7360N).
>>
>> Here's my main question, do I need to get a special card to run a cable
>> from my computer to the Fax?
>>
>> Is it better to get a dedicated fax machine to use, or are there
>> specialized fax machines for Fax over IP?
>>
>> Thanks guys and gals,
>>
>> Tim
>>
>> --
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From bdfoster at endigotech.com  Fri Apr 13 23:17:27 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 13 Apr 2012 15:17:27 -0400
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CAMwZ-tJZoXOHuemkHYbBAvihBKwuAUjOjB0PHAx5qzdUGPm-gg@mail.gmail.com>
References: <CAE3mkNDJie83wpu3r9xOmVDGvcBnvUDUgsubkBPhtNuMghASZg@mail.gmail.com>
	<CBADBBAB.311D1%krice@freeswitch.org>
	<CAMwZ-tJZoXOHuemkHYbBAvihBKwuAUjOjB0PHAx5qzdUGPm-gg@mail.gmail.com>
Message-ID: <CAKV-jFxfFvSXJht0Gvb-Yc+yTXNj7f79RzjHNO1hPb_R7bvp7w@mail.gmail.com>

On Fri, Apr 13, 2012 at 1:23 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Apr 13, 2012 at 10:16 AM, Ken Rice <krice at freeswitch.org> wrote:
> > Data stored in there is used for various things like show calls and show
> > channels so that we don't have to lock memory structures just to show you
> > what calls are active and what state they are in. (this was a lessoned
> > learned from years of seeing "avoiding deadlock" from a simple show me
> the
> > calls.
>
> I don't want to get all nostalgic, but I miss those days :)
>

Heh... I don't.



>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
-------------- next part --------------
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From wesleyakio at tuntscorp.com  Sat Apr 14 01:06:33 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Fri, 13 Apr 2012 18:06:33 -0300
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
Message-ID: <CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>

Sorry to hop into this conversation, but sice the subject is fax...

Brian,

You recomended  api_hangup_hook and that is currently what I'm using, I
tried execute_on_fax_success as it seemed more apropriate but I can't get
it to work properly. I'm still debuging but it seems to work only with
small faxes...

Have you tried execute_on_fax_success? Did it work for you?

Wesley Akio
TuntsCorp.com


On Fri, Apr 13, 2012 at 3:29 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent on
> network conditions and carrier support and implementation.
>
> With that out of the way, the best place to start doing fax using your FS
> machine (only) would be here:
>
> Place this in your public context (maybe it shouldn't stay there, but this
> is an extremely easy way to mess around with faxing):
>
> <extension name="fax_receive">
>   <condition field="context" expression="public"/>
>   <condition field="destination_number" expression="^9978$">
>     <action application="answer"/>
>     <action application="playback" data="silence_stream://2000"/>
>     <action application="set" data="fax_enable_t38_request=true"/>
>     <action application="set" data="fax_enable_t38=true"/>
>     <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> If you want to run a script after the fax is recieved, put this line in
> before the "rxfax":
>
>     <action application="set" data="api_hangup_hook=system <scriptname>
> <arg1> <arg2> <argN>"/>
>
> Start with receiving faxes, and go from there. Sending faxes requires some
> extra work, but not much. For more information about faxing, please visit
> http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax
>
> Obviously if you have any questions, please do not hesitate to ask!
>
> -BDF
>
>
> On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>
>>  You've raised some good questions, and good points.
>>
>> This model does have an RJ-11, and I believe you are correct.  I don't
>> have an ATA yet, but I'll get the ATA model that you have (Grandstream
>> HT286 ATA).
>>
>> 1 - I'll do a dedicated fax number
>> 1.1 - Each fax machine will have its own number
>> 2 -  As it stands, I'm going to be running this over VoIP
>>
>> As far as faxing via FS w/o the machine goes, I wasn't really aware of
>> that.  I think I knew it, but it didn't click.  After I get the ATA setup
>> working, I think I'll set up another version where FS is handling it w/o a
>> machine.  That way I'll have both types of experience under my belt.
>>
>> Actually, as I think about it more.. it probably makes more sense to set
>> up the environment where the machine isn't present first (as I don't have
>> the ATA on hand now)...
>>
>> Thanks!
>>
>>
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>>
>>
>> On 4/13/2012 12:51 PM, Brian Foster wrote:
>>
>> I've looked at that model, and it doesn't actually say this but I'm
>> assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
>> my faxing using SpanDSP, but basically what you would need is an ATA that
>> supports T.38, as well as a provider that supports T.38. Flowroute seems to
>> be pretty reliable (that's who I use). I have a fax machine that I've
>> played around with here, and I'm using a Grandstream HT286 ATA to connect
>> it to FreeSWITCH.
>>
>> Having said all of that, it's highly dependent on your setup. Questions:
>>
>> 1. Are you going to have dedicated fax numbers?
>> 1.1. If so, will each fax machine have it's own fax number?
>> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>>
>> If there's any other details you feel are related, please mention those
>> too. Realize that FS can do faxing on it's own without the need for a fax
>> machine. Most, however, decide not to go down that route mostly because the
>> fax machine is an interface most are comfortable with. You can even do
>> outbound faxes on a fax machine, and get FS to handle incoming faxes and
>> email them, store them, and even print them to a network printer.
>>
>> We're more than happy to help!
>>
>> -BDF
>>
>> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>>
>>> Hey all,
>>>
>>> I know that there is a good amount of information out there on how to do
>>> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>>>
>>> I'm currently trying to figure out the process by which I would need to
>>> do this.  I'll detail my setup, and ask questions.  I've read
>>> documentation, but I'm still missing something.
>>>
>>> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
>>> and I'm the only user.  I'm using FlowRoute, and I have a couple of
>>> unused numbers.  Pretty simple, nothing out of the ordinary (except
>>> maybe the VM part?).
>>>
>>> At some point in the next six months or so, I'll be getting a couple of
>>> dedicated servers here for various things, and I'll begin moving my VM
>>> servers from my main computer to those, as actual hardware hosts.
>>>
>>> So my fax machine is built into an "All-in-one" Network
>>> Printer/Scanner/Fax (Brother MFC-7360N).
>>>
>>> Here's my main question, do I need to get a special card to run a cable
>>> from my computer to the Fax?
>>>
>>> Is it better to get a dedicated fax machine to use, or are there
>>> specialized fax machines for Fax over IP?
>>>
>>> Thanks guys and gals,
>>>
>>> Tim
>>>
>>> --
>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>> Quarry Worker's Creed
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Sat Apr 14 01:14:39 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 13 Apr 2012 17:14:39 -0400
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
Message-ID: <CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>

I have not, I actually learned about it today doing research for this
particular problem. I'm interested in trying it, though. Yes, it seems more
appropriate because you can also do execute_on_fax_failure and do something
else when it fails i.e. delete the file that was just created, send an
email out saying there was a failed fax, etc. The reason why I hadn't
mentioned is was just because I had not tried it, and I didn't want to
recommend it until I had tried it out. I'll do some research though.

-BDF

On Fri, Apr 13, 2012 at 5:06 PM, Wesley Akio <wesleyakio at tuntscorp.com>wrote:

> Sorry to hop into this conversation, but sice the subject is fax...
>
> Brian,
>
> You recomended  api_hangup_hook and that is currently what I'm using, I
> tried execute_on_fax_success as it seemed more apropriate but I can't get
> it to work properly. I'm still debuging but it seems to work only with
> small faxes...
>
> Have you tried execute_on_fax_success? Did it work for you?
>
> Wesley Akio
> TuntsCorp.com
>
>
>
> On Fri, Apr 13, 2012 at 3:29 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent on
>> network conditions and carrier support and implementation.
>>
>> With that out of the way, the best place to start doing fax using your FS
>> machine (only) would be here:
>>
>> Place this in your public context (maybe it shouldn't stay there, but
>> this is an extremely easy way to mess around with faxing):
>>
>> <extension name="fax_receive">
>>   <condition field="context" expression="public"/>
>>   <condition field="destination_number" expression="^9978$">
>>     <action application="answer"/>
>>     <action application="playback" data="silence_stream://2000"/>
>>     <action application="set" data="fax_enable_t38_request=true"/>
>>     <action application="set" data="fax_enable_t38=true"/>
>>     <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
>>     <action application="hangup"/>
>>   </condition>
>> </extension>
>>
>> If you want to run a script after the fax is recieved, put this line in
>> before the "rxfax":
>>
>>     <action application="set" data="api_hangup_hook=system <scriptname>
>> <arg1> <arg2> <argN>"/>
>>
>> Start with receiving faxes, and go from there. Sending faxes requires
>> some extra work, but not much. For more information about faxing, please
>> visit http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax
>>
>> Obviously if you have any questions, please do not hesitate to ask!
>>
>> -BDF
>>
>>
>> On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>>
>>>  You've raised some good questions, and good points.
>>>
>>> This model does have an RJ-11, and I believe you are correct.  I don't
>>> have an ATA yet, but I'll get the ATA model that you have (Grandstream
>>> HT286 ATA).
>>>
>>> 1 - I'll do a dedicated fax number
>>> 1.1 - Each fax machine will have its own number
>>> 2 -  As it stands, I'm going to be running this over VoIP
>>>
>>> As far as faxing via FS w/o the machine goes, I wasn't really aware of
>>> that.  I think I knew it, but it didn't click.  After I get the ATA setup
>>> working, I think I'll set up another version where FS is handling it w/o a
>>> machine.  That way I'll have both types of experience under my belt.
>>>
>>> Actually, as I think about it more.. it probably makes more sense to set
>>> up the environment where the machine isn't present first (as I don't have
>>> the ATA on hand now)...
>>>
>>> Thanks!
>>>
>>>
>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>> Quarry Worker's Creed
>>>
>>>
>>> On 4/13/2012 12:51 PM, Brian Foster wrote:
>>>
>>> I've looked at that model, and it doesn't actually say this but I'm
>>> assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
>>> my faxing using SpanDSP, but basically what you would need is an ATA that
>>> supports T.38, as well as a provider that supports T.38. Flowroute seems to
>>> be pretty reliable (that's who I use). I have a fax machine that I've
>>> played around with here, and I'm using a Grandstream HT286 ATA to connect
>>> it to FreeSWITCH.
>>>
>>> Having said all of that, it's highly dependent on your setup. Questions:
>>>
>>> 1. Are you going to have dedicated fax numbers?
>>> 1.1. If so, will each fax machine have it's own fax number?
>>> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>>>
>>> If there's any other details you feel are related, please mention those
>>> too. Realize that FS can do faxing on it's own without the need for a fax
>>> machine. Most, however, decide not to go down that route mostly because the
>>> fax machine is an interface most are comfortable with. You can even do
>>> outbound faxes on a fax machine, and get FS to handle incoming faxes and
>>> email them, store them, and even print them to a network printer.
>>>
>>> We're more than happy to help!
>>>
>>> -BDF
>>>
>>> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>>>
>>>> Hey all,
>>>>
>>>> I know that there is a good amount of information out there on how to do
>>>> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>>>>
>>>> I'm currently trying to figure out the process by which I would need to
>>>> do this.  I'll detail my setup, and ask questions.  I've read
>>>> documentation, but I'm still missing something.
>>>>
>>>> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
>>>> and I'm the only user.  I'm using FlowRoute, and I have a couple of
>>>> unused numbers.  Pretty simple, nothing out of the ordinary (except
>>>> maybe the VM part?).
>>>>
>>>> At some point in the next six months or so, I'll be getting a couple of
>>>> dedicated servers here for various things, and I'll begin moving my VM
>>>> servers from my main computer to those, as actual hardware hosts.
>>>>
>>>> So my fax machine is built into an "All-in-one" Network
>>>> Printer/Scanner/Fax (Brother MFC-7360N).
>>>>
>>>> Here's my main question, do I need to get a special card to run a cable
>>>> from my computer to the Fax?
>>>>
>>>> Is it better to get a dedicated fax machine to use, or are there
>>>> specialized fax machines for Fax over IP?
>>>>
>>>> Thanks guys and gals,
>>>>
>>>> Tim
>>>>
>>>> --
>>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>>> Quarry Worker's Creed
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>> you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be intercepted,
>>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>> The sender therefore does not accept liability for any errors or omissions
>>> in the contents of this message, which arise as a result of e-mail
>>> transmission. If verification is required please request a hard-copy
>>> version.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From anthony.minessale at gmail.com  Sat Apr 14 02:00:05 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 13 Apr 2012 17:00:05 -0500
Subject: [Freeswitch-users] Doing basic arithmetic in the xml dialplan
 without a language
In-Reply-To: <CAC-LwPP0arzUH-ayqESMKvOwp=nrvMZFmvuLjaqHf-rz2ri8ng@mail.gmail.com>
References: <CAGTy0Gyw-H0VdJAZ5wa5g9zbKHqV=fBG=HHD4iY_n6yOU0bbpA@mail.gmail.com>
	<CAC-LwPP0arzUH-ayqESMKvOwp=nrvMZFmvuLjaqHf-rz2ri8ng@mail.gmail.com>
Message-ID: <CAKbxfG8oXGmjrAM7VNrWDt1BEWP-CvPDii9G6qDETbhC9q4Suw@mail.gmail.com>

Also FYI to make the API as variable syntax lighter on the eyes if you
don't need the () you can omit them:

<action application="set" data="inc=${expr ${inc}+1}"/>




On Fri, Apr 13, 2012 at 6:52 AM, Vik Killa <vipkilla at gmail.com> wrote:
> This is an example of adding 1 to the 'inc' variable. If 'inc' is not
> defined, it will become 1. If it is 1, it will become 2, etc...
> <action application="set" data="inc=${expr(${inc}+1)}"/>
>
> On Fri, Apr 13, 2012 at 7:24 AM, Herman Griffin
> <herman.griffin at gmail.com> wrote:
>> Is there a way to do basic arithmetic within the dialplan without running a
>> mod_language script. For instance, to increment a counter on every call. Or
>> to use the modula or division.
>>
>> Herman
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From wesleyakio at tuntscorp.com  Sat Apr 14 03:54:49 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Fri, 13 Apr 2012 20:54:49 -0300
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
	<CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
Message-ID: <CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>

Please let me know if you figure it out! I only have ISDN on my production
boxes and sip providers over here are not really into T.38 therefore it's a
bit complicated to debug...

Wesley Akio

On Fri, Apr 13, 2012 at 6:14 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> I have not, I actually learned about it today doing research for this
> particular problem. I'm interested in trying it, though. Yes, it seems more
> appropriate because you can also do execute_on_fax_failure and do something
> else when it fails i.e. delete the file that was just created, send an
> email out saying there was a failed fax, etc. The reason why I hadn't
> mentioned is was just because I had not tried it, and I didn't want to
> recommend it until I had tried it out. I'll do some research though.
>
> -BDF
>
>
> On Fri, Apr 13, 2012 at 5:06 PM, Wesley Akio <wesleyakio at tuntscorp.com>wrote:
>
>> Sorry to hop into this conversation, but sice the subject is fax...
>>
>> Brian,
>>
>> You recomended  api_hangup_hook and that is currently what I'm using, I
>> tried execute_on_fax_success as it seemed more apropriate but I can't get
>> it to work properly. I'm still debuging but it seems to work only with
>> small faxes...
>>
>> Have you tried execute_on_fax_success? Did it work for you?
>>
>> Wesley Akio
>> TuntsCorp.com
>>
>>
>>
>> On Fri, Apr 13, 2012 at 3:29 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>>
>>> Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent on
>>> network conditions and carrier support and implementation.
>>>
>>> With that out of the way, the best place to start doing fax using your
>>> FS machine (only) would be here:
>>>
>>> Place this in your public context (maybe it shouldn't stay there, but
>>> this is an extremely easy way to mess around with faxing):
>>>
>>> <extension name="fax_receive">
>>>   <condition field="context" expression="public"/>
>>>   <condition field="destination_number" expression="^9978$">
>>>     <action application="answer"/>
>>>     <action application="playback" data="silence_stream://2000"/>
>>>     <action application="set" data="fax_enable_t38_request=true"/>
>>>     <action application="set" data="fax_enable_t38=true"/>
>>>     <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
>>>     <action application="hangup"/>
>>>   </condition>
>>> </extension>
>>>
>>> If you want to run a script after the fax is recieved, put this line in
>>> before the "rxfax":
>>>
>>>     <action application="set" data="api_hangup_hook=system <scriptname>
>>> <arg1> <arg2> <argN>"/>
>>>
>>> Start with receiving faxes, and go from there. Sending faxes requires
>>> some extra work, but not much. For more information about faxing, please
>>> visit http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax
>>>
>>> Obviously if you have any questions, please do not hesitate to ask!
>>>
>>> -BDF
>>>
>>>
>>> On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>>>
>>>>  You've raised some good questions, and good points.
>>>>
>>>> This model does have an RJ-11, and I believe you are correct.  I don't
>>>> have an ATA yet, but I'll get the ATA model that you have (Grandstream
>>>> HT286 ATA).
>>>>
>>>> 1 - I'll do a dedicated fax number
>>>> 1.1 - Each fax machine will have its own number
>>>> 2 -  As it stands, I'm going to be running this over VoIP
>>>>
>>>> As far as faxing via FS w/o the machine goes, I wasn't really aware of
>>>> that.  I think I knew it, but it didn't click.  After I get the ATA setup
>>>> working, I think I'll set up another version where FS is handling it w/o a
>>>> machine.  That way I'll have both types of experience under my belt.
>>>>
>>>> Actually, as I think about it more.. it probably makes more sense to
>>>> set up the environment where the machine isn't present first (as I don't
>>>> have the ATA on hand now)...
>>>>
>>>> Thanks!
>>>>
>>>>
>>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>>> Quarry Worker's Creed
>>>>
>>>>
>>>> On 4/13/2012 12:51 PM, Brian Foster wrote:
>>>>
>>>> I've looked at that model, and it doesn't actually say this but I'm
>>>> assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
>>>> my faxing using SpanDSP, but basically what you would need is an ATA that
>>>> supports T.38, as well as a provider that supports T.38. Flowroute seems to
>>>> be pretty reliable (that's who I use). I have a fax machine that I've
>>>> played around with here, and I'm using a Grandstream HT286 ATA to connect
>>>> it to FreeSWITCH.
>>>>
>>>> Having said all of that, it's highly dependent on your setup. Questions:
>>>>
>>>> 1. Are you going to have dedicated fax numbers?
>>>> 1.1. If so, will each fax machine have it's own fax number?
>>>> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>>>>
>>>> If there's any other details you feel are related, please mention those
>>>> too. Realize that FS can do faxing on it's own without the need for a fax
>>>> machine. Most, however, decide not to go down that route mostly because the
>>>> fax machine is an interface most are comfortable with. You can even do
>>>> outbound faxes on a fax machine, and get FS to handle incoming faxes and
>>>> email them, store them, and even print them to a network printer.
>>>>
>>>> We're more than happy to help!
>>>>
>>>> -BDF
>>>>
>>>> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com
>>>> > wrote:
>>>>
>>>>> Hey all,
>>>>>
>>>>> I know that there is a good amount of information out there on how to
>>>>> do
>>>>> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>>>>>
>>>>> I'm currently trying to figure out the process by which I would need to
>>>>> do this.  I'll detail my setup, and ask questions.  I've read
>>>>> documentation, but I'm still missing something.
>>>>>
>>>>> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
>>>>> and I'm the only user.  I'm using FlowRoute, and I have a couple of
>>>>> unused numbers.  Pretty simple, nothing out of the ordinary (except
>>>>> maybe the VM part?).
>>>>>
>>>>> At some point in the next six months or so, I'll be getting a couple of
>>>>> dedicated servers here for various things, and I'll begin moving my VM
>>>>> servers from my main computer to those, as actual hardware hosts.
>>>>>
>>>>> So my fax machine is built into an "All-in-one" Network
>>>>> Printer/Scanner/Fax (Brother MFC-7360N).
>>>>>
>>>>> Here's my main question, do I need to get a special card to run a cable
>>>>> from my computer to the Fax?
>>>>>
>>>>> Is it better to get a dedicated fax machine to use, or are there
>>>>> specialized fax machines for Fax over IP?
>>>>>
>>>>> Thanks guys and gals,
>>>>>
>>>>> Tim
>>>>>
>>>>> --
>>>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>>>> Quarry Worker's Creed
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Brian D. Foster
>>>> Endigo Computer LLC
>>>> Email: bdfoster at endigotech.com
>>>> Phone: 317-800-7876
>>>> Indianapolis, Indiana, USA
>>>>
>>>> This message contains confidential information and is intended for
>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message
>>>> header. If you are not the intended recipient you are notified that
>>>> disclosing, copying, distributing or taking any action in reliance on the
>>>> contents of this information is strictly prohibited. E-mail transmission
>>>> cannot be guaranteed to be secure or error-free as information could be
>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or
>>>> contain viruses. The sender therefore does not accept liability for any
>>>> errors or omissions in the contents of this message, which arise as a
>>>> result of e-mail transmission. If verification is required please request a
>>>> hard-copy version.
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>
>>>> 
>>>>
>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>> you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be intercepted,
>>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>> The sender therefore does not accept liability for any errors or omissions
>>> in the contents of this message, which arise as a result of e-mail
>>> transmission. If verification is required please request a hard-copy
>>> version.
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mitch.capper at gmail.com  Sat Apr 14 05:45:59 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Fri, 13 Apr 2012 18:45:59 -0700
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F88642F.9010706@gmail.com>
References: <4F88642F.9010706@gmail.com>
Message-ID: <CALn233qpT1zCdV73gyR6R8ShzFbV16K5rAyUgKWJv5rX0uy5_Q@mail.gmail.com>

So you probably don't want to use the machine in Fax mode but rather
Scan mode.  Get a favorite mode setup to scan to a TIF and then fax
through FS directly (although even if you scan in as a TIF you may
want a script to properly resize it per the wiki).  In addition
running fax in a VM or over lower quality internet connections is
going to hurt your success rates.  With that said a good internet
connection + flowroute can get you some very good results just follow
the wiki.

~Mitch


From jmsucasas at me.com  Sat Apr 14 05:11:30 2012
From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto)
Date: Sat, 14 Apr 2012 03:11:30 +0200
Subject: [Freeswitch-users] Authenticate external calls problem
Message-ID: <C06636AC-6827-4AC2-B8E1-CEDD6848D290@me.com>

Hello,

   I have installed freeSwitch 1.1beta from the git repository (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000).  My SIP provider (Netelip) is defined in conf/directory/default/example.conf.xml:

<include>
  <user id="$${default_provider}">
    <gateways>
      <gateway name="$${default_provider}">
	<param name="username" value="$${default_provider_username}"/>
	<param name="password" value="$${default_provider_password}"/>
	<param name="from-user" value="$${default_provider_username}"/>
	<param name="from-domain" value="$${default_provider_from_domain}"/>
	<param name="expire-seconds" value="600"/>
	<param name="register" value="$${default_provider_register}"/>
	<param name="retry-seconds" value="30"/>
	<param name="extension" value="$${default_provider_contact}"/>
	<param name="auth-calls" value="false"/>
	<!--<param name="contact-params" value="domain_name=$${domain}"/>-->
	<param name="context" value="public"/>
      </gateway>
    </gateways>
    <params>
      <param name="password" value="$${default_provider_password}"/>
    </params>
  </user>
</include>

	and doesn't authenticate inbound calls (insecure=invite in asterisk language). When I receive an inbound call,  I receive the following error:

freeswitch at internal> 
recv 1000 bytes from udp/[194.140.135.80]:5060 at 23:23:10.163377:
   ------------------------------------------------------------------------
   INVITE sip:34810101189 at sukmad.dyndns-home.com SIP/2.0
   Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport
   From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
   To: <sip:34810101189 at sukmad.dyndns-home.com>
   Contact: <sip:916495242 at 194.140.135.80>
   Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
   CSeq: 102 INVITE
   User-Agent: Netelip
   Max-Forwards: 70
   Date: Fri, 13 Apr 2012 23:23:51 GMT
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
   Supported: replaces
   Content-Type: application/sdp
   Content-Length: 425
   
   v=0
   o=root 1249 1249 IN IP4 194.140.135.80
   s=session
   c=IN IP4 194.140.135.80
   t=0 0
   m=audio 16474 RTP/AVP 8 0 97 3 18 4 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:97 iLBC/8000
   a=fmtp:97 mode=30
   a=rtpmap:3 GSM/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:4 G723/8000
   a=fmtp:4 annexa=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   a=sendrecv
   ------------------------------------------------------------------------
send 382 bytes to udp/[194.140.135.80]:5060 at 23:23:10.164409:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
   From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
   To: <sip:34810101189 at sukmad.dyndns-home.com>
   Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000
   Content-Length: 0
   
   ------------------------------------------------------------------------
2012-04-14 01:23:10.162331 [DEBUG] sofia.c:7598 IP 194.140.135.80 Rejected by acl "domains". Falling back to Digest auth.
send 871 bytes to udp/[194.140.135.80]:5060 at 23:23:10.166026:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
   From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
   To: <sip:34810101189 at sukmad.dyndns-home.com>;tag=SaBHcvmm7evNF
   Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
   CSeq: 102 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="194.140.135.80", nonce="bc1571d4-0957-4609-a887-7c3359b9ae9c", algorithm=MD5, qop="auth"
   Content-Length: 0

	I have read in forums that the solution to this problem is to add the IP to the acl.conf.xml, like this:

    <list name="domains" default="deny">
      <!-- domain= is special it scans the domain from the directory to build the ACL -->
      <node type="allow" domain="$${domain}"/>
      <!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
      <node type="allow" cidr="194.140.135.80/24"/> 
    </list>

	But my SIP provider have a dynamic IP, and it's behind a NAT.  If my SIP provider will use any IP, ?Where can I do this without enabling all IPs to enter calls in my system?


Thanks in advance,

Regards.
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From vfclists at gmail.com  Sat Apr 14 18:11:11 2012
From: vfclists at gmail.com (Frank Church)
Date: Sat, 14 Apr 2012 15:11:11 +0100
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CBADBBAB.311D1%krice@freeswitch.org>
References: <CAE3mkNDJie83wpu3r9xOmVDGvcBnvUDUgsubkBPhtNuMghASZg@mail.gmail.com>
	<CBADBBAB.311D1%krice@freeswitch.org>
Message-ID: <CAE3mkNALxP5pVW9Ct3qb7KJy-k6HAhpD7vjnXGHAR1vgAUO+pA@mail.gmail.com>

On 13 April 2012 17:16, Ken Rice <krice at freeswitch.org> wrote:

> The files in the db directory are used for a variety of things...
> Most of the files are for storing runtime information that can be nuked
> across reboots the exception to that is the voicemail db... If you nuke it
> you lose all your voicemail metadata effectively resetting all the
> voicemail
> boxes.
>
> Data stored in there is used for various things like show calls and show
> channels so that we don't have to lock memory structures just to show you
> what calls are active and what state they are in. (this was a lessoned
> learned from years of seeing "avoiding deadlock" from a simple show me the
> calls.
>
> As far as using an alternate database yes you can... See the wiki for core
> odbc docs. This applies not just to the core but to several other modules
> individually.
>
> Also as a side note, if you are not using the voicemail module, you can
> mount the db directory on a ram drive. This has been shown to increse
> performance and reduce disk IO a fair bit...
>
>
>
> On 4/13/12 4:38 AM, "Frank Church" <vfclists at gmail.com> wrote:
>
> > I have been getting an occasional problem when freeswitch fails to
> > start correctly. I googled around and found out that its was caused by
> > the corruption of files in the db directory.
> >
> > What runtime related information is stored in those files? Are they
> > useful for some logging or analytical value?
> >
> > Can an alternate database system be used for them?
>
>
>
>

Is there a way to detect if the start up failure is due to those fails, so
that you can delete them and restart Freeswitch?

Can you check something like asterisks 'core show channels' etc from it,
something like call progress, duration etc of live calls, CDR etc?

What format are they in, sqlite3?




_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From anita.hall at simmortel.com  Sat Apr 14 19:51:54 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Sat, 14 Apr 2012 21:21:54 +0530
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
Message-ID: <CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>

Could he use mod_odbc_query instead of xml curl to directly get his
dialplan from a DB, thus removing the webserver from the picture
altogether. I have not tried this but it looks better or am I missing
something ?

http://wiki.freeswitch.org/wiki/Mod_odbc_query

regards,
Anita



On Fri, Apr 13, 2012 at 10:52 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Fri, Apr 13, 2012 at 6:37 AM, huseyin kalyoncu <hkalyoncu at gmail.com>
> wrote:
> > we are currently have two fs boxes load balanced by osips.
> > in each fs we have dynamic dialplan with mod_xml_curl using php & apache
> &
> > mysql. this conf is doing just fine with current load (about 20 cps) but
> with the
> > increasing of incoming traffic, it shows some performance issues.
> > we want to increase our call throughput.
>
> You don't really give us enough to go on. A few questions that I have
> when I read this...
>
> * Are the DB and HTTP servers running on stand alone servers, or are
> they running on the FS server?
> * What kind of hardware are they each running on? RAM, CPU, disks?
> * What kind of load do the servers have when you see performance issues?
> * What have you done to optimize your web stack (other HTTP servers
> etc)? Are you caching where you can?
> * Have you analyzed your SQL to find any slow queries?
>
> Now, I don't expect you to answer all of these in this email thread,
> but I'll bet if you reviewed them, you'd be able to bump your
> performance without having to revisit your current approach.
>
>
> > i searched through fs site and mailing list and came up with following
> > options:
>
> I'll comment on this generically, but I can't really say what each
> approach will (or will not) do for your current situation.
>
>
> > 1) using lua or (python?) to serve dialplan instead of mod_xml_curl
>
> This works. Lua is lighter so it may give you better performance than
> Python (this is a non-issue if you're using Python in a long-living
> process with event socket or mod_xml_curl). Regardless of languages,
> this approach has a drawback in that processing is done on the same
> server that's running FreeSWITCH.
>
>
> > 2) writing a dialplan module (something like mod_xml_curl) in c which
> will do
> > all the db lookups etc..
>
> That seems a little extreme. Many people have scaled way up without
> resorting to that approach.
>
>
> > 3) using mod_erlang_event in outbound mode & spawning several erlang
> > workers to do db lookups etc..
>
> This is about the same as #4 (give or take some Erlang magic).
>
>
> > 4) using mod_event_socket in outbound mode & making db lookups on a
> > different server.
>
> This is a fine approach when controlling the call, but you still need
> some basic dialplan to get the calls going where they need to go. I
> don't see it as the right way to solve your scaling problem, but it
> might be part of the overall solution.
>
>
> It sounds like you might be giving up on mod_xml_curl too soon.
> Scaling HTTP is a well known problem with lots of tools available to
> help you. When developing for mod_xml_curl, try returning the simplest
> bit of dialplan you can and do all the work (logic, DB, etc.) on the
> HTTP server. And don't forget that in *any* scaling scenario, you will
> need to bring hardware, *real* hardware :)
>
> Good luck and let us know how you end up solving the problem.
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Sat Apr 14 20:02:41 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 14 Apr 2012 11:02:41 -0500
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
	<CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>
Message-ID: <CAKbxfG9q9nee7qbFVnWqZDgvvQpSHrrhqzB4Nj7XuN3QUspjQA@mail.gmail.com>

If you really want the fastest dialplan possible, write a new dialplan
module in C.
You only have to implement 1 function, a dialplan hunt function that
gets data from the session and builds and returns an extension object
which is basically just a list of app/args.  You can glue in any other
stuff you want from there.

The dialplan implementations we carry in tree tend to offer a blended
balance of performance and flexibility/extensibility to serve as both
a demo and a practical on-ramp.





On Sat, Apr 14, 2012 at 10:51 AM, Anita Hall <anita.hall at simmortel.com> wrote:
> Could he use mod_odbc_query instead of xml curl to directly get his dialplan
> from a DB, thus removing the webserver from the picture altogether. I have
> not tried this but it looks better or am I missing something ?
>
> http://wiki.freeswitch.org/wiki/Mod_odbc_query
>
> regards,
> Anita
>
>
>
>
> On Fri, Apr 13, 2012 at 10:52 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>>
>> On Fri, Apr 13, 2012 at 6:37 AM, huseyin kalyoncu <hkalyoncu at gmail.com>
>> wrote:
>> > we are currently have two fs boxes load balanced by osips.
>> > in each fs we have dynamic dialplan with mod_xml_curl using php & apache
>> > &
>> > mysql. this conf is doing just fine with current load (about 20 cps) but
>> > with the
>> > increasing of incoming traffic, it shows some performance issues.
>> > we want to increase our call throughput.
>>
>> You don't really give us enough to go on. A few questions that I have
>> when I read this...
>>
>> * Are the DB and HTTP servers running on stand alone servers, or are
>> they running on the FS server?
>> * What kind of hardware are they each running on? RAM, CPU, disks?
>> * What kind of load do the servers have when you see performance issues?
>> * What have you done to optimize your web stack (other HTTP servers
>> etc)? Are you caching where you can?
>> * Have you analyzed your SQL to find any slow queries?
>>
>> Now, I don't expect you to answer all of these in this email thread,
>> but I'll bet if you reviewed them, you'd be able to bump your
>> performance without having to revisit your current approach.
>>
>>
>> > i searched through fs site and mailing list and came up with following
>> > options:
>>
>> I'll comment on this generically, but I can't really say what each
>> approach will (or will not) do for your current situation.
>>
>>
>> > 1) using lua or (python?) to serve dialplan instead of mod_xml_curl
>>
>> This works. Lua is lighter so it may give you better performance than
>> Python (this is a non-issue if you're using Python in a long-living
>> process with event socket or mod_xml_curl). Regardless of languages,
>> this approach has a drawback in that processing is done on the same
>> server that's running FreeSWITCH.
>>
>>
>> > 2) writing a dialplan module (something like mod_xml_curl) in c which
>> > will do
>> > all the db lookups etc..
>>
>> That seems a little extreme. Many people have scaled way up without
>> resorting to that approach.
>>
>>
>> > 3) using mod_erlang_event in outbound mode & spawning several erlang
>> > workers to do db lookups etc..
>>
>> This is about the same as #4 (give or take some Erlang magic).
>>
>>
>> > 4) using mod_event_socket in outbound mode & making db lookups on a
>> > different server.
>>
>> This is a fine approach when controlling the call, but you still need
>> some basic dialplan to get the calls going where they need to go. I
>> don't see it as the right way to solve your scaling problem, but it
>> might be part of the overall solution.
>>
>>
>> It sounds like you might be giving up on mod_xml_curl too soon.
>> Scaling HTTP is a well known problem with lots of tools available to
>> help you. When developing for mod_xml_curl, try returning the simplest
>> bit of dialplan you can and do all the work (logic, DB, etc.) on the
>> HTTP server. And don't forget that in *any* scaling scenario, you will
>> need to bring hardware, *real* hardware :)
>>
>> Good luck and let us know how you end up solving the problem.
>>
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From krice at freeswitch.org  Sat Apr 14 20:19:30 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 14 Apr 2012 11:19:30 -0500
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CAE3mkNALxP5pVW9Ct3qb7KJy-k6HAhpD7vjnXGHAR1vgAUO+pA@mail.gmail.com>
Message-ID: <CBAF0DC2.312AB%krice@freeswitch.org>

If you arent using Voicemail you can just delete them prior to startup every
time...

However, I would say if you are getting this routinely this is just symptom
of a larger problem... And its like treating just the pain from the bullet
in your foot with advil and not treating the bullet wound...

The question should be why are these corrupting...

I run 100s of 1000s (of not millions) of sessions per day on freeswitch with
recent versions of the code and do not see this issue, so I?m left wondering
if theres some reason this is happening like hard killing of the box or some
other bug...

Iirc it is sqlite3, but we have a defacto fork of SQLite running due to a
variety of things we had to re-arrange in the code. Standard SQLite tools
should still allow you to play with the .db files tho

K


On 4/14/12 9:11 AM, "Frank Church" <vfclists at gmail.com> wrote:

> Is there a way to detect if the start up failure is due to those fails, so
> that you can delete them and restart Freeswitch?
> 
> Can you check something like asterisks 'core show channels' etc from it,
> something like call progress, duration etc of live calls, CDR etc?
> 
> What format are they in, sqlite3?
> 
> 
> 

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From ocset at the800group.com  Sat Apr 14 20:48:26 2012
From: ocset at the800group.com (ocset)
Date: Sun, 15 Apr 2012 00:48:26 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
References: <4F87D110.50103@the800group.com>
	<CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
Message-ID: <4F89AA5A.3040402@the800group.com>

Thanks Nandy

I am not using the GSM device in this test and have tried changing the 
disconnect signalling without any luck. When I hang up the phones, 
nothing is recorder in the Freeswitch log (log level 7). Is there a way 
to trap the disconnect signal by writing a "lua" script or would that 
info already be written to the log if it was being received?.

What I don't understand is that when I use the SIP gateway for the 
outgoing leg of the call, then the disconnect signal from the two phones 
is received. But when I use the PSTN line as the outgoing leg, the 
disconnect signal is not received from either phone?

Anyone got any suggestions how I would debug this?

Thanks again for your help.
O


On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
> check the line supervision signalling of your GSM and the GXW4104. 
> there are several disconnect signalling used: current disconnect, 
> polarity reversal or busy tones. your FXO port must be set accordingly.
>
> On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Hi
>
>     I have successfully managed to get FreeSwitch to bridge an
>     incoming call
>     on a PSTN line to an external mobile phone using a second PSTN
>     line with
>     the help of a GXW4104 FSO gateway.
>
>     I am now having a serious issue where the call never gets terminated
>     when the parties hang up. The only way to end the call is to either
>     shut-down FreeSwitch or turn off the GXW4104. If I bridge the call
>     to a
>     SIP gateway (PennyTel) instead of the second PSTN line, then it
>     works as
>     expected when both parties hang up.
>
>     Here is what I have tested
>
>     1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external
>     mobile phone -- call end when both parties hang up.
>     2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external
>     mobile phone -- call never end, even when both parties hang up.
>
>     Is there a parameter I should be setting to help solve this issue or
>     some code that I should implement to test for this situation?
>
>     I have also noticed that with test scenario 1 (PSTN line + SIP
>     Gateway),
>     if the person being called does not end the call on their side, then
>     that leg of the bridge remains active.
>
>     ps. The second PSTN line is not a real copper line. It is a PennyTel
>     VOIP line that is managed by the modem and connected to the
>     GXW4104 via
>     phone cable. So, as far as the GXW4104 is concerned, it is just
>     another
>     POTS line. I don't have a second PSTN phone line so can't test any
>     other
>     way.
>
>     Thanks in advance
>     O.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anita.hall at simmortel.com  Sat Apr 14 23:06:43 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Sun, 15 Apr 2012 00:36:43 +0530
Subject: [Freeswitch-users] Nuance mod_unimrcp ASR
Message-ID: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>

Hi

I am making a FreeSWITCH application for ASR using Nuance Speech Server 5.0
and mod_unimrcp. What are my options ?

regards,
Anita
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From gcd at i.ph  Sun Apr 15 05:01:02 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Sun, 15 Apr 2012 09:01:02 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <4F89AA5A.3040402@the800group.com>
References: <4F87D110.50103@the800group.com>
	<CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
	<4F89AA5A.3040402@the800group.com>
Message-ID: <CAHKqMb7u4E5+9ow_eJ=f5_X_hiVW4iD3sBz5rhqZ6fMBYtaQGQ@mail.gmail.com>

GSM device or not, just determine what line disconnect signalling is used
in each port. the GWX gateway must be set to detect this. otherwise,
there's no way FS can hangup the call. this signal must be present when
using the SIP gateway.

also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up
first.

On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset at the800group.com> wrote:

>  Thanks Nandy
>
> I am not using the GSM device in this test and have tried changing the
> disconnect signalling without any luck. When I hang up the phones, nothing
> is recorder in the Freeswitch log (log level 7). Is there a way to trap the
> disconnect signal by writing a "lua" script or would that info already be
> written to the log if it was being received?.
>
> What I don't understand is that when I use the SIP gateway for the
> outgoing leg of the call, then the disconnect signal from the two phones is
> received. But when I use the PSTN line as the outgoing leg, the disconnect
> signal is not received from either phone?
>
> Anyone got any suggestions how I would debug this?
>
> Thanks again for your help.
> O
>
>
>
> On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
>
> check the line supervision signalling of your GSM and the GXW4104. there
> are several disconnect signalling used: current disconnect, polarity
> reversal or busy tones. your FXO port must be set accordingly.
>
>  On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com> wrote:
>
>> Hi
>>
>> I have successfully managed to get FreeSwitch to bridge an incoming call
>> on a PSTN line to an external mobile phone using a second PSTN line with
>> the help of a GXW4104 FSO gateway.
>>
>> I am now having a serious issue where the call never gets terminated
>> when the parties hang up. The only way to end the call is to either
>> shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a
>> SIP gateway (PennyTel) instead of the second PSTN line, then it works as
>> expected when both parties hang up.
>>
>> Here is what I have tested
>>
>> 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external
>> mobile phone -- call end when both parties hang up.
>> 2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external
>> mobile phone -- call never end, even when both parties hang up.
>>
>> Is there a parameter I should be setting to help solve this issue or
>> some code that I should implement to test for this situation?
>>
>> I have also noticed that with test scenario 1 (PSTN line + SIP Gateway),
>> if the person being called does not end the call on their side, then
>> that leg of the bridge remains active.
>>
>> ps. The second PSTN line is not a real copper line. It is a PennyTel
>> VOIP line that is managed by the modem and connected to the GXW4104 via
>> phone cable. So, as far as the GXW4104 is concerned, it is just another
>> POTS line. I don't have a second PSTN phone line so can't test any other
>> way.
>>
>> Thanks in advance
>> O.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From ifoundthetao at gmail.com  Sun Apr 15 07:12:03 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Sat, 14 Apr 2012 22:12:03 -0500
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
Message-ID: <4F8A3C83.4020109@gmail.com>

Brain (:,

I went the route of setting it up inside of FS first.  The code from the 
Wiki worked perfectly!  It was interesting seeing it come in via fs_cli, 
watching the process, seeing the file-size change, then getting the all 
clear when it officially completed.  I received my first fax.

Now.. on to sending faxes!

Thanks for your help

'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 4/13/2012 1:29 PM, Brian Foster wrote:
> Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent 
> on network conditions and carrier support and implementation.
>
> With that out of the way, the best place to start doing fax using your 
> FS machine (only) would be here:
>
> Place this in your public context (maybe it shouldn't stay there, but 
> this is an extremely easy way to mess around with faxing):
>
> <extension name="fax_receive">
> <condition field="context" expression="public"/>
> <condition field="destination_number" expression="^9978$">
> <action application="answer"/>
> <action application="playback" data="silence_stream://2000"/>
> <action application="set" data="fax_enable_t38_request=true"/>
> <action application="set" data="fax_enable_t38=true"/>
> <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
> If you want to run a script after the fax is recieved, put this line 
> in before the "rxfax":
>
> <action application="set" data="api_hangup_hook=system <scriptname> 
> <arg1> <arg2> <argN>"/>
>
> Start with receiving faxes, and go from there. Sending faxes requires 
> some extra work, but not much. For more information about faxing, 
> please visit http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax
>
> Obviously if you have any questions, please do not hesitate to ask!
>
> -BDF
>
> On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton 
> <ifoundthetao at gmail.com <mailto:ifoundthetao at gmail.com>> wrote:
>
>     You've raised some good questions, and good points.
>
>     This model does have an RJ-11, and I believe you are correct.  I
>     don't have an ATA yet, but I'll get the ATA model that you have
>     (Grandstream HT286 ATA).
>
>     1 - I'll do a dedicated fax number
>     1.1 - Each fax machine will have its own number
>     2 -  As it stands, I'm going to be running this over VoIP
>
>     As far as faxing via FS w/o the machine goes, I wasn't really
>     aware of that.  I think I knew it, but it didn't click.  After I
>     get the ATA setup working, I think I'll set up another version
>     where FS is handling it w/o a machine.  That way I'll have both
>     types of experience under my belt.
>
>     Actually, as I think about it more.. it probably makes more sense
>     to set up the environment where the machine isn't present first
>     (as I don't have the ATA on hand now)...
>
>     Thanks!
>
>
>     'We who cut mere stones must always be envisioning cathedrals.'
>     Quarry Worker's Creed
>
>
>     On 4/13/2012 12:51 PM, Brian Foster wrote:
>>     I've looked at that model, and it doesn't actually say this but
>>     I'm assuming you connect an RJ-11 to the fax modem inside the
>>     all-in-one. I do my faxing using SpanDSP, but basically what you
>>     would need is an ATA that supports T.38, as well as a provider
>>     that supports T.38. Flowroute seems to be pretty reliable (that's
>>     who I use). I have a fax machine that I've played around with
>>     here, and I'm using a Grandstream HT286 ATA to connect it to
>>     FreeSWITCH.
>>
>>     Having said all of that, it's highly dependent on your setup.
>>     Questions:
>>
>>     1. Are you going to have dedicated fax numbers?
>>     1.1. If so, will each fax machine have it's own fax number?
>>     2. Are you going to have this running over VoIP or TDM
>>     (PRI/BRI/etc.)?
>>
>>     If there's any other details you feel are related, please mention
>>     those too. Realize that FS can do faxing on it's own without the
>>     need for a fax machine. Most, however, decide not to go down that
>>     route mostly because the fax machine is an interface most are
>>     comfortable with. You can even do outbound faxes on a fax
>>     machine, and get FS to handle incoming faxes and email them,
>>     store them, and even print them to a network printer.
>>
>>     We're more than happy to help!
>>
>>     -BDF
>>
>>     On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton
>>     <ifoundthetao at gmail.com <mailto:ifoundthetao at gmail.com>> wrote:
>>
>>         Hey all,
>>
>>         I know that there is a good amount of information out there
>>         on how to do
>>         faxing via FreeSWITCH, but I'm still finding myself a bit
>>         confused.
>>
>>         I'm currently trying to figure out the process by which I
>>         would need to
>>         do this.  I'll detail my setup, and ask questions.  I've read
>>         documentation, but I'm still missing something.
>>
>>         Okay, I'm currently running FreeSWITCH in a CentOS VM, with
>>         4GB of RAM,
>>         and I'm the only user.  I'm using FlowRoute, and I have a
>>         couple of
>>         unused numbers.  Pretty simple, nothing out of the ordinary
>>         (except
>>         maybe the VM part?).
>>
>>         At some point in the next six months or so, I'll be getting a
>>         couple of
>>         dedicated servers here for various things, and I'll begin
>>         moving my VM
>>         servers from my main computer to those, as actual hardware hosts.
>>
>>         So my fax machine is built into an "All-in-one" Network
>>         Printer/Scanner/Fax (Brother MFC-7360N).
>>
>>         Here's my main question, do I need to get a special card to
>>         run a cable
>>         from my computer to the Fax?
>>
>>         Is it better to get a dedicated fax machine to use, or are there
>>         specialized fax machines for Fax over IP?
>>
>>         Thanks guys and gals,
>>
>>         Tim
>>
>>         --
>>         'We who cut mere stones must always be envisioning cathedrals.'
>>         Quarry Worker's Creed
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Brian D. Foster
>>     Endigo Computer LLC
>>     Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>>     Phone: 317-800-7876 <tel:317-800-7876>
>>     Indianapolis, Indiana, USA
>>
>>     This message contains confidential information and is intended
>>     for those listed in the "To:", "CC:", and/or "BCC:" fields of the
>>     message header. If you are not the intended recipient you are
>>     notified that disclosing, copying, distributing or taking any
>>     action in reliance on the contents of this information is
>>     strictly prohibited. E-mail transmission cannot be guaranteed to
>>     be secure or error-free as information could be intercepted,
>>     corrupted, lost, destroyed, arrive late or incomplete, or contain
>>     viruses. The sender therefore does not accept liability for any
>>     errors or omissions in the contents of this message, which arise
>>     as a result of e-mail transmission. If verification is required
>>     please request a hard-copy version.
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for 
> those listed in the "To:", "CC:", and/or "BCC:" fields of the message 
> header. If you are not the intended recipient you are notified that 
> disclosing, copying, distributing or taking any action in reliance on 
> the contents of this information is strictly prohibited. E-mail 
> transmission cannot be guaranteed to be secure or error-free as 
> information could be intercepted, corrupted, lost, destroyed, arrive 
> late or incomplete, or contain viruses. The sender therefore does not 
> accept liability for any errors or omissions in the contents of this 
> message, which arise as a result of e-mail transmission. If 
> verification is required please request a hard-copy version.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Sun Apr 15 08:11:13 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sun, 15 Apr 2012 08:11:13 +0400
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F8A3C83.4020109@gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<4F8A3C83.4020109@gmail.com>
Message-ID: <CANBhyPBpft46KEPuX4rs0YAnLYvpqd2hWchx1iMmzSHnkmx3fQ@mail.gmail.com>

Cool. The next step would be to send fax using ITSP (the quote from
FlowRoute twitter: "Did you know we support T.38 on all domestic and
international routes?") which support T.38 faxing over IP. Just add this
little snippet to your extension for outgoing calls (don't forget to set
T.38 @ your ATA) and you good to go:

<action application="export" data="t38_passthru=true"/>
http://wiki.freeswitch.org/wiki/T.38

15 ?????? 2012 ?. 7:12 ???????????? Timothy Bolton
<ifoundthetao at gmail.com>???????:

>  Brain (:,
>
> I went the route of setting it up inside of FS first.  The code from the
> Wiki worked perfectly!  It was interesting seeing it come in via fs_cli,
> watching the process, seeing the file-size change, then getting the all
> clear when it officially completed.  I received my first fax.
>
> Now.. on to sending faxes!
>
> Thanks for your help
>
>
> 'We who cut mere stones must always be envisioning cathedrals.'
> Quarry Worker's Creed
>
>
> On 4/13/2012 1:29 PM, Brian Foster wrote:
>
> Disclaimer: Faxing over VoIP can be a crapshoot. It's very dependent on
> network conditions and carrier support and implementation.
>
> With that out of the way, the best place to start doing fax using your FS
> machine (only) would be here:
>
> Place this in your public context (maybe it shouldn't stay there, but this
> is an extremely easy way to mess around with faxing):
>
> <extension name="fax_receive">
>   <condition field="context" expression="public"/>
>   <condition field="destination_number" expression="^9978$">
>     <action application="answer"/>
>     <action application="playback" data="silence_stream://2000"/>
>     <action application="set" data="fax_enable_t38_request=true"/>
>     <action application="set" data="fax_enable_t38=true"/>
>     <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> If you want to run a script after the fax is recieved, put this line in
> before the "rxfax":
>
>     <action application="set" data="api_hangup_hook=system <scriptname>
> <arg1> <arg2> <argN>"/>
>
> Start with receiving faxes, and go from there. Sending faxes requires some
> extra work, but not much. For more information about faxing, please visit
> http://wiki.freeswitch.org/wiki/Mod_spandsp#Fax
>
> Obviously if you have any questions, please do not hesitate to ask!
>
> -BDF
>
> On Fri, Apr 13, 2012 at 2:09 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>
>>  You've raised some good questions, and good points.
>>
>> This model does have an RJ-11, and I believe you are correct.  I don't
>> have an ATA yet, but I'll get the ATA model that you have (Grandstream
>> HT286 ATA).
>>
>> 1 - I'll do a dedicated fax number
>> 1.1 - Each fax machine will have its own number
>> 2 -  As it stands, I'm going to be running this over VoIP
>>
>> As far as faxing via FS w/o the machine goes, I wasn't really aware of
>> that.  I think I knew it, but it didn't click.  After I get the ATA setup
>> working, I think I'll set up another version where FS is handling it w/o a
>> machine.  That way I'll have both types of experience under my belt.
>>
>> Actually, as I think about it more.. it probably makes more sense to set
>> up the environment where the machine isn't present first (as I don't have
>> the ATA on hand now)...
>>
>> Thanks!
>>
>>
>> 'We who cut mere stones must always be envisioning cathedrals.'
>> Quarry Worker's Creed
>>
>>
>>   On 4/13/2012 12:51 PM, Brian Foster wrote:
>>
>> I've looked at that model, and it doesn't actually say this but I'm
>> assuming you connect an RJ-11 to the fax modem inside the all-in-one. I do
>> my faxing using SpanDSP, but basically what you would need is an ATA that
>> supports T.38, as well as a provider that supports T.38. Flowroute seems to
>> be pretty reliable (that's who I use). I have a fax machine that I've
>> played around with here, and I'm using a Grandstream HT286 ATA to connect
>> it to FreeSWITCH.
>>
>> Having said all of that, it's highly dependent on your setup. Questions:
>>
>> 1. Are you going to have dedicated fax numbers?
>> 1.1. If so, will each fax machine have it's own fax number?
>> 2. Are you going to have this running over VoIP or TDM (PRI/BRI/etc.)?
>>
>> If there's any other details you feel are related, please mention those
>> too. Realize that FS can do faxing on it's own without the need for a fax
>> machine. Most, however, decide not to go down that route mostly because the
>> fax machine is an interface most are comfortable with. You can even do
>> outbound faxes on a fax machine, and get FS to handle incoming faxes and
>> email them, store them, and even print them to a network printer.
>>
>> We're more than happy to help!
>>
>> -BDF
>>
>> On Fri, Apr 13, 2012 at 1:36 PM, Timothy Bolton <ifoundthetao at gmail.com>wrote:
>>
>>> Hey all,
>>>
>>> I know that there is a good amount of information out there on how to do
>>> faxing via FreeSWITCH, but I'm still finding myself a bit confused.
>>>
>>> I'm currently trying to figure out the process by which I would need to
>>> do this.  I'll detail my setup, and ask questions.  I've read
>>> documentation, but I'm still missing something.
>>>
>>> Okay, I'm currently running FreeSWITCH in a CentOS VM, with 4GB of RAM,
>>> and I'm the only user.  I'm using FlowRoute, and I have a couple of
>>> unused numbers.  Pretty simple, nothing out of the ordinary (except
>>> maybe the VM part?).
>>>
>>> At some point in the next six months or so, I'll be getting a couple of
>>> dedicated servers here for various things, and I'll begin moving my VM
>>> servers from my main computer to those, as actual hardware hosts.
>>>
>>> So my fax machine is built into an "All-in-one" Network
>>> Printer/Scanner/Fax (Brother MFC-7360N).
>>>
>>> Here's my main question, do I need to get a special card to run a cable
>>> from my computer to the Fax?
>>>
>>> Is it better to get a dedicated fax machine to use, or are there
>>> specialized fax machines for Fax over IP?
>>>
>>> Thanks guys and gals,
>>>
>>> Tim
>>>
>>> --
>>> 'We who cut mere stones must always be envisioning cathedrals.'
>>> Quarry Worker's Creed
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fieldpeak at gmail.com  Sun Apr 15 09:07:13 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 15 Apr 2012 13:07:13 +0800
Subject: [Freeswitch-users] media bypass for mod_callcenter
Message-ID: <CAOTmbMw5eg6s+GGKAjODAFmgK2Q=3Wwbkj=rGC2M91LWk5GA4w@mail.gmail.com>

Hi Guys,

call flow: pstn->media GW->FS->mod_callcenter->hear music->agent
i need media go between media GW and agent, bypass FS, can anyone help
advise how to configure, thanks.

-- 
Regards,
Charles
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From shahzad.bhatti at g-r-v.com  Sun Apr 15 10:02:55 2012
From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti)
Date: Sun, 15 Apr 2012 11:02:55 +0500
Subject: [Freeswitch-users] LUASQL Installation Problem!
Message-ID: <CAN+NqUZiu9bygk2iZwGjRxL_xe8sPLL7KnxmyDq9bW7ujE-jWw@mail.gmail.com>

AOA,


i tried a lot to install LUASQL but have errors in it .


i follow wiki for it http://wiki.freeswitch.org/wiki/Installing_LuaSQL


here is my *config* file data

[root at localhost luasql-2.1.1]# vim config
# Driver (leave uncommented ONLY the line with the name of the driver)
T= mysql
#T= oci8
#T= odbc
#T= postgres
#T= sqlite
#T=sqlite3

# Installation directories

# Default prefix
PREFIX = /usr/local

# System's libraries directory (where binary libraries are installed)
LUA_LIBDIR= $(PREFIX)/lib/lua/5.1

# System's lua directory (where Lua libraries are installed)
LUA_DIR= $(PREFIX)/share/lua/5.1

# Lua includes directory
#LUA_INC= $(PREFIX)/include
LUA_INC= /usr/src/freeswitch/src/mod/languages/mod_lua/lua

# Lua version number (first and second digits of target version)
LUA_VERSION_NUM= 501

# OS dependent
LIB_OPTION= -shared #for Linux
#LIB_OPTION= -bundle -undefined dynamic_lookup #for MacOS X

LIBNAME= $T.so
COMPAT_DIR= ../compat/src

# Compilation parameters
# Driver specific
####### MySQL
DRIVER_LIBS= -L/usr/local/mysql/lib -lmysqlclient -lz
DRIVER_INCS= -I/usr/local/mysql/include
######## Oracle OCI8
#DRIVER_LIBS= -L/home/oracle/OraHome1/lib -lz -lclntsh
#DRIVER_INCS= -I/home/oracle/OraHome1/rdbms/demo
-I/home/oracle/OraHome1/rdbms/public
######## PostgreSQL
#DRIVER_LIBS= -L/usr/local/pgsql/lib -lpq
#DRIVER_INCS= -I/usr/local/pgsql/include
######## SQLite
#DRIVER_LIBS= -lsqlite
#DRIVER_INCS=
######## SQLite3
#DRIVER_LIBS= -L/opt/local/lib -lsqlite3
#DRIVER_INCS= -I/opt/local/include
######## ODBC
#DRIVER_LIBS= -L/usr/local/lib -lodbc
#DRIVER_INCS= -DUNIXODBC -I/usr/local/include

WARN= -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic
INCS= -I$(LUA_INC)
CFLAGS= -O2 $(WARN) -I$(COMPAT_DIR) $(DRIVER_INCS) $(INCS) $(DEFS)
CC= gcc

# $Id: config,v 1.8 2007/10/27 22:55:27 carregal Exp $



but get the error :

my output is as followos:

[root at localhost luasql-2.1.1]# make
gcc -O2 -Wall -Wmissing-prototypes -Wmissing-declarations -ansi -pedantic
-I../compat/src  -I/usr/src/freeswitch/src/mod/languages/mod_lua/lua    -c
-o src/ls_mysql.o src/ls_mysql.c
src/ls_mysql.c:19:19: error: mysql.h: No such file or directory
src/ls_mysql.c:73: error: expected specifier-qualifier-list before ?MYSQL?
src/ls_mysql.c:81: error: expected specifier-qualifier-list before
?MYSQL_RES?
src/ls_mysql.c:148: warning: ISO C forbids forward references to ?enum?
types
src/ls_mysql.c:148: warning: ?enum enum_field_types? declared inside
parameter list
src/ls_mysql.c:148: warning: its scope is only this definition or
declaration, which is probably not what you want
src/ls_mysql.c:148: error: parameter 1 (?type?) has incomplete type
src/ls_mysql.c: In function ?getcolumntype?:
src/ls_mysql.c:151: error: ?FIELD_TYPE_VAR_STRING? undeclared (first use in
this function)
src/ls_mysql.c:151: error: (Each undeclared identifier is reported only once
src/ls_mysql.c:151: error: for each function it appears in.)
src/ls_mysql.c:151: error: ?FIELD_TYPE_STRING? undeclared (first use in
this function)
src/ls_mysql.c:153: error: ?FIELD_TYPE_DECIMAL? undeclared (first use in
this function)
src/ls_mysql.c:153: error: ?FIELD_TYPE_SHORT? undeclared (first use in this
function)
src/ls_mysql.c:153: error: ?FIELD_TYPE_LONG? undeclared (first use in this
function)
src/ls_mysql.c:154: error: ?FIELD_TYPE_FLOAT? undeclared (first use in this
function)
src/ls_mysql.c:154: error: ?FIELD_TYPE_DOUBLE? undeclared (first use in
this function)
src/ls_mysql.c:154: error: ?FIELD_TYPE_LONGLONG? undeclared (first use in
this function)
src/ls_mysql.c:155: error: ?FIELD_TYPE_INT24? undeclared (first use in this
function)
src/ls_mysql.c:155: error: ?FIELD_TYPE_YEAR? undeclared (first use in this
function)
src/ls_mysql.c:155: error: ?FIELD_TYPE_TINY? undeclared (first use in this
function)
src/ls_mysql.c:157: error: ?FIELD_TYPE_TINY_BLOB? undeclared (first use in
this function)
src/ls_mysql.c:157: error: ?FIELD_TYPE_MEDIUM_BLOB? undeclared (first use
in this function)
src/ls_mysql.c:158: error: ?FIELD_TYPE_LONG_BLOB? undeclared (first use in
this function)
src/ls_mysql.c:158: error: ?FIELD_TYPE_BLOB? undeclared (first use in this
function)
src/ls_mysql.c:160: error: ?FIELD_TYPE_DATE? undeclared (first use in this
function)
src/ls_mysql.c:160: error: ?FIELD_TYPE_NEWDATE? undeclared (first use in
this function)
src/ls_mysql.c:162: error: ?FIELD_TYPE_DATETIME? undeclared (first use in
this function)
src/ls_mysql.c:164: error: ?FIELD_TYPE_TIME? undeclared (first use in this
function)
src/ls_mysql.c:166: error: ?FIELD_TYPE_TIMESTAMP? undeclared (first use in
this function)
src/ls_mysql.c:168: error: ?FIELD_TYPE_ENUM? undeclared (first use in this
function)
src/ls_mysql.c:168: error: ?FIELD_TYPE_SET? undeclared (first use in this
function)
src/ls_mysql.c:170: error: ?FIELD_TYPE_NULL? undeclared (first use in this
function)
src/ls_mysql.c: In function ?create_colinfo?:
src/ls_mysql.c:182: error: ?MYSQL_FIELD? undeclared (first use in this
function)
src/ls_mysql.c:182: error: ?fields? undeclared (first use in this function)
src/ls_mysql.c:183: warning: ISO C90 forbids mixed declarations and code
src/ls_mysql.c:185: warning: implicit declaration of function
?mysql_fetch_fields?
src/ls_mysql.c:185: error: ?cur_data? has no member named ?my_res?
src/ls_mysql.c:191: error: type of formal parameter 1 is incomplete
src/ls_mysql.c: In function ?cur_fetch?:
src/ls_mysql.c:206: error: ?MYSQL_RES? undeclared (first use in this
function)
src/ls_mysql.c:206: error: ?res? undeclared (first use in this function)
src/ls_mysql.c:206: error: ?cur_data? has no member named ?my_res?
src/ls_mysql.c:207: warning: ISO C90 forbids mixed declarations and code
src/ls_mysql.c:208: error: ?MYSQL_ROW? undeclared (first use in this
function)
src/ls_mysql.c:208: error: expected ?;? before ?row?
src/ls_mysql.c:209: error: ?row? undeclared (first use in this function)
src/ls_mysql.c:213: warning: implicit declaration of function
?mysql_fetch_lengths?
src/ls_mysql.c: In function ?cur_close?:
src/ls_mysql.c:269: warning: implicit declaration of function
?mysql_free_result?
src/ls_mysql.c:269: error: ?cur_data? has no member named ?my_res?
src/ls_mysql.c: In function ?cur_numrows?:
src/ls_mysql.c:319: warning: implicit declaration of function
?mysql_num_rows?
src/ls_mysql.c:319: error: ?cur_data? has no member named ?my_res?
src/ls_mysql.c: At top level:
src/ls_mysql.c:327: error: expected declaration specifiers or ?...? before
?MYSQL_RES?
src/ls_mysql.c: In function ?create_cursor?:
src/ls_mysql.c:337: error: ?cur_data? has no member named ?my_res?
src/ls_mysql.c:337: error: ?result? undeclared (first use in this function)
src/ls_mysql.c: In function ?conn_close?:
src/ls_mysql.c:359: warning: implicit declaration of function ?mysql_close?
src/ls_mysql.c:359: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c: In function ?conn_execute?:
src/ls_mysql.c:374: warning: implicit declaration of function
?mysql_real_query?
src/ls_mysql.c:374: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:376: warning: implicit declaration of function ?mysql_error?
src/ls_mysql.c:376: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:379: error: ?MYSQL_RES? undeclared (first use in this
function)
src/ls_mysql.c:379: error: ?res? undeclared (first use in this function)
src/ls_mysql.c:379: warning: implicit declaration of function
?mysql_store_result?
src/ls_mysql.c:379: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:380: warning: implicit declaration of function
?mysql_field_count?
src/ls_mysql.c:380: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:380: warning: ISO C90 forbids mixed declarations and code
src/ls_mysql.c:383: error: too many arguments to function ?create_cursor?
src/ls_mysql.c:388: warning: implicit declaration of function
?mysql_affected_rows?
src/ls_mysql.c:388: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:392: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c: In function ?conn_commit?:
src/ls_mysql.c:403: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c: In function ?conn_rollback?:
src/ls_mysql.c:413: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c: In function ?conn_setautocommit?:
src/ls_mysql.c:424: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:427: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c: At top level:
src/ls_mysql.c:437: error: expected declaration specifiers or ?...? before
?MYSQL?
src/ls_mysql.c: In function ?create_connection?:
src/ls_mysql.c:444: error: ?conn_data? has no member named ?my_conn?
src/ls_mysql.c:444: error: ?my_conn? undeclared (first use in this function)
src/ls_mysql.c: In function ?env_connect?:
src/ls_mysql.c:462: error: ?MYSQL? undeclared (first use in this function)
src/ls_mysql.c:462: error: ?conn? undeclared (first use in this function)
src/ls_mysql.c:466: warning: implicit declaration of function ?mysql_init?
src/ls_mysql.c:470: warning: implicit declaration of function
?mysql_real_connect?
src/ls_mysql.c:478: error: too many arguments to function
?create_connection?
src/ls_mysql.c: In function ?luaopen_luasql_mysql?:
src/ls_mysql.c:557: error: expected ?)? before ?MYSQL_SERVER_VERSION?
src/ls_mysql.c:557: error: too few arguments to function ?lua_pushlstring?
make: *** [src/ls_mysql.o] Error 1
[root at localhost luasql-2.1.1]#




please tell me the reason of the error:


Regards

Shahzad Bhatti
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From anton.vazir at gmail.com  Sun Apr 15 15:11:37 2012
From: anton.vazir at gmail.com (Anton VG)
Date: Sun, 15 Apr 2012 16:11:37 +0500
Subject: [Freeswitch-users] creating a conference for 2 separate parked
	UUIDs, please help
Message-ID: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>

Guys,

I need to create a conference for 2 UUID's , which already exists and
parked. How do I create a new, empty conference and then send those 2
UUIDs there?


From ocset at the800group.com  Sun Apr 15 20:38:49 2012
From: ocset at the800group.com (ocset)
Date: Mon, 16 Apr 2012 00:38:49 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <CAHKqMb7u4E5+9ow_eJ=f5_X_hiVW4iD3sBz5rhqZ6fMBYtaQGQ@mail.gmail.com>
References: <4F87D110.50103@the800group.com>
	<CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
	<4F89AA5A.3040402@the800group.com>
	<CAHKqMb7u4E5+9ow_eJ=f5_X_hiVW4iD3sBz5rhqZ6fMBYtaQGQ@mail.gmail.com>
Message-ID: <4F8AF999.1000405@the800group.com>

Hi Nandy

Thanks for you help.

Just to answer you question first. It makes not difference who hangs up 
first, the calls never terminate.

I have enabled "Polarity Reversal" on the GXW4104 and that has solved 
the problem. What I don't get is why this would work. I thought Polarity 
Reversal is old school and how would my mobile phone be doing this or is 
this being done by the mobile phone provider/carrier?

I now have "Enable Current Disconnect", "Enable Tone Disconnect" and 
"Enable Polarity Reversal" set for good measure even though the "Reverse 
Polarity" was the one that seemed to fix the problem.

Regards
O.

On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
> GSM device or not, just determine what line disconnect signalling is 
> used in each port. the GWX gateway must be set to detect this. 
> otherwise, there's no way FS can hangup the call. this signal must be 
> present when using the SIP gateway.
>
> also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up 
> first.
>
> On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset at the800group.com 
> <mailto:ocset at the800group.com>> wrote:
>
>     Thanks Nandy
>
>     I am not using the GSM device in this test and have tried changing
>     the disconnect signalling without any luck. When I hang up the
>     phones, nothing is recorder in the Freeswitch log (log level 7).
>     Is there a way to trap the disconnect signal by writing a "lua"
>     script or would that info already be written to the log if it was
>     being received?.
>
>     What I don't understand is that when I use the SIP gateway for the
>     outgoing leg of the call, then the disconnect signal from the two
>     phones is received. But when I use the PSTN line as the outgoing
>     leg, the disconnect signal is not received from either phone?
>
>     Anyone got any suggestions how I would debug this?
>
>     Thanks again for your help.
>     O
>
>
>
>     On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
>>     check the line supervision signalling of your GSM and the
>>     GXW4104. there are several disconnect signalling used: current
>>     disconnect, polarity reversal or busy tones. your FXO port must
>>     be set accordingly.
>>
>>     On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com
>>     <mailto:ocset at the800group.com>> wrote:
>>
>>         Hi
>>
>>         I have successfully managed to get FreeSwitch to bridge an
>>         incoming call
>>         on a PSTN line to an external mobile phone using a second
>>         PSTN line with
>>         the help of a GXW4104 FSO gateway.
>>
>>         I am now having a serious issue where the call never gets
>>         terminated
>>         when the parties hang up. The only way to end the call is to
>>         either
>>         shut-down FreeSwitch or turn off the GXW4104. If I bridge the
>>         call to a
>>         SIP gateway (PennyTel) instead of the second PSTN line, then
>>         it works as
>>         expected when both parties hang up.
>>
>>         Here is what I have tested
>>
>>         1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway ->
>>         external
>>         mobile phone -- call end when both parties hang up.
>>         2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) ->
>>         external
>>         mobile phone -- call never end, even when both parties hang up.
>>
>>         Is there a parameter I should be setting to help solve this
>>         issue or
>>         some code that I should implement to test for this situation?
>>
>>         I have also noticed that with test scenario 1 (PSTN line +
>>         SIP Gateway),
>>         if the person being called does not end the call on their
>>         side, then
>>         that leg of the bridge remains active.
>>
>>         ps. The second PSTN line is not a real copper line. It is a
>>         PennyTel
>>         VOIP line that is managed by the modem and connected to the
>>         GXW4104 via
>>         phone cable. So, as far as the GXW4104 is concerned, it is
>>         just another
>>         POTS line. I don't have a second PSTN phone line so can't
>>         test any other
>>         way.
>>
>>         Thanks in advance
>>         O.
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Sun Apr 15 20:46:26 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 15 Apr 2012 12:46:26 -0400
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <4F8AF999.1000405@the800group.com>
References: <4F87D110.50103@the800group.com>
	<CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
	<4F89AA5A.3040402@the800group.com>
	<CAHKqMb7u4E5+9ow_eJ=f5_X_hiVW4iD3sBz5rhqZ6fMBYtaQGQ@mail.gmail.com>
	<4F8AF999.1000405@the800group.com>
Message-ID: <CAKV-jFx1e1QZxmZBuBQQB-xR_EoUY8_9GSLP6s4hNbCrJO4b6g@mail.gmail.com>

It is what your telephone company's equipment does, not your mobile phone
provider.
On Apr 15, 2012 12:39 PM, "ocset" <ocset at the800group.com> wrote:

>  Hi Nandy
>
> Thanks for you help.
>
> Just to answer you question first. It makes not difference who hangs up
> first, the calls never terminate.
>
> I have enabled "Polarity Reversal" on the GXW4104 and that has solved the
> problem. What I don't get is why this would work. I thought Polarity
> Reversal is old school and how would my mobile phone be doing this or is
> this being done by the mobile phone provider/carrier?
>
> I now have "Enable Current Disconnect", "Enable Tone Disconnect" and
> "Enable Polarity Reversal" set for good measure even though the "Reverse
> Polarity" was the one that seemed to fix the problem.
>
> Regards
> O.
>
> On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
>
> GSM device or not, just determine what line disconnect signalling is used
> in each port. the GWX gateway must be set to detect this. otherwise,
> there's no way FS can hangup the call. this signal must be present when
> using the SIP gateway.
>
> also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up
> first.
>
> On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset at the800group.com> wrote:
>
>>  Thanks Nandy
>>
>> I am not using the GSM device in this test and have tried changing the
>> disconnect signalling without any luck. When I hang up the phones, nothing
>> is recorder in the Freeswitch log (log level 7). Is there a way to trap the
>> disconnect signal by writing a "lua" script or would that info already be
>> written to the log if it was being received?.
>>
>> What I don't understand is that when I use the SIP gateway for the
>> outgoing leg of the call, then the disconnect signal from the two phones is
>> received. But when I use the PSTN line as the outgoing leg, the disconnect
>> signal is not received from either phone?
>>
>> Anyone got any suggestions how I would debug this?
>>
>> Thanks again for your help.
>> O
>>
>>
>>
>> On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
>>
>> check the line supervision signalling of your GSM and the GXW4104. there
>> are several disconnect signalling used: current disconnect, polarity
>> reversal or busy tones. your FXO port must be set accordingly.
>>
>>  On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com> wrote:
>>
>>> Hi
>>>
>>> I have successfully managed to get FreeSwitch to bridge an incoming call
>>> on a PSTN line to an external mobile phone using a second PSTN line with
>>> the help of a GXW4104 FSO gateway.
>>>
>>> I am now having a serious issue where the call never gets terminated
>>> when the parties hang up. The only way to end the call is to either
>>> shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a
>>> SIP gateway (PennyTel) instead of the second PSTN line, then it works as
>>> expected when both parties hang up.
>>>
>>> Here is what I have tested
>>>
>>> 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external
>>> mobile phone -- call end when both parties hang up.
>>> 2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external
>>> mobile phone -- call never end, even when both parties hang up.
>>>
>>> Is there a parameter I should be setting to help solve this issue or
>>> some code that I should implement to test for this situation?
>>>
>>> I have also noticed that with test scenario 1 (PSTN line + SIP Gateway),
>>> if the person being called does not end the call on their side, then
>>> that leg of the bridge remains active.
>>>
>>> ps. The second PSTN line is not a real copper line. It is a PennyTel
>>> VOIP line that is managed by the modem and connected to the GXW4104 via
>>> phone cable. So, as far as the GXW4104 is concerned, it is just another
>>> POTS line. I don't have a second PSTN phone line so can't test any other
>>> way.
>>>
>>> Thanks in advance
>>> O.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From manjiri05_deshpande at yahoo.co.in  Mon Apr 16 00:45:26 2012
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Mon, 16 Apr 2012 04:45:26 +0800 (SGT)
Subject: [Freeswitch-users] Fwd:
Message-ID: <1334522726.44368.BPMail_high_noncarrier@web192506.mail.sg3.yahoo.com>


http://cityspots.ro/play.php?jvuwys=481&anjgenax=11


From azza.miled at gmail.com  Mon Apr 16 02:26:03 2012
From: azza.miled at gmail.com (azza miled)
Date: Mon, 16 Apr 2012 00:26:03 +0200
Subject: [Freeswitch-users] Integrating FreeWITCH and Opensips
Message-ID: <CAGQdkzTgfKEZT5+vEvTFot0-XL0Y4adbWwK4kvzOOf_8Q5etOg@mail.gmail.com>

Hi!
I want to use FreeSWITCH to handle IVR for my opensips's users.I have no
idea where to start.A detailed tutoriel will be helpful.
 Can any one help me, please?
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From bdfoster at endigotech.com  Mon Apr 16 02:34:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 15 Apr 2012 18:34:12 -0400
Subject: [Freeswitch-users] Integrating FreeWITCH and Opensips
In-Reply-To: <CAGQdkzTgfKEZT5+vEvTFot0-XL0Y4adbWwK4kvzOOf_8Q5etOg@mail.gmail.com>
References: <CAGQdkzTgfKEZT5+vEvTFot0-XL0Y4adbWwK4kvzOOf_8Q5etOg@mail.gmail.com>
Message-ID: <CAKV-jFxGLBybV9tbXmwvKGYMCQECHOxXQEoAzxpyKK1iF8O97g@mail.gmail.com>

There are tutorials on how to make an IVR on the wiki, see
http://wiki.freeswitch.org and search for IVR.
On Apr 15, 2012 6:27 PM, "azza miled" <azza.miled at gmail.com> wrote:

> Hi!
> I want to use FreeSWITCH to handle IVR for my opensips's users.I have no
> idea where to start.A detailed tutoriel will be helpful.
>  Can any one help me, please?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gcd at i.ph  Mon Apr 16 03:01:52 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 16 Apr 2012 07:01:52 +0800
Subject: [Freeswitch-users] Bridged call not ending/terminating
In-Reply-To: <CAKV-jFx1e1QZxmZBuBQQB-xR_EoUY8_9GSLP6s4hNbCrJO4b6g@mail.gmail.com>
References: <4F87D110.50103@the800group.com>
	<CAHKqMb7AShH1BsNyLOf-WmztbagrXhQL_HgGa1-AAWANKpA5TQ@mail.gmail.com>
	<4F89AA5A.3040402@the800group.com>
	<CAHKqMb7u4E5+9ow_eJ=f5_X_hiVW4iD3sBz5rhqZ6fMBYtaQGQ@mail.gmail.com>
	<4F8AF999.1000405@the800group.com>
	<CAKV-jFx1e1QZxmZBuBQQB-xR_EoUY8_9GSLP6s4hNbCrJO4b6g@mail.gmail.com>
Message-ID: <CAHKqMb7yVF0-BmYW-QeU-P2khxGj9m0V8TTCQ=cat2f2BW7-XA@mail.gmail.com>

nice to know it worked! old school? there are billions of POTS lines still
using it although they're feed by digital exchanges:-)

On Mon, Apr 16, 2012 at 12:46 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> It is what your telephone company's equipment does, not your mobile phone
> provider.
> On Apr 15, 2012 12:39 PM, "ocset" <ocset at the800group.com> wrote:
>
>>  Hi Nandy
>>
>> Thanks for you help.
>>
>> Just to answer you question first. It makes not difference who hangs up
>> first, the calls never terminate.
>>
>> I have enabled "Polarity Reversal" on the GXW4104 and that has solved the
>> problem. What I don't get is why this would work. I thought Polarity
>> Reversal is old school and how would my mobile phone be doing this or is
>> this being done by the mobile phone provider/carrier?
>>
>> I now have "Enable Current Disconnect", "Enable Tone Disconnect" and
>> "Enable Polarity Reversal" set for good measure even though the "Reverse
>> Polarity" was the one that seemed to fix the problem.
>>
>> Regards
>> O.
>>
>> On 04/15/2012 09:01 AM, Nandy Dagondon wrote:
>>
>> GSM device or not, just determine what line disconnect signalling is used
>> in each port. the GWX gateway must be set to detect this. otherwise,
>> there's no way FS can hangup the call. this signal must be present when
>> using the SIP gateway.
>>
>> also test these 2 scenarios - 1) FXO1 hangs up first; 2) FXO2 hangs up
>> first.
>>
>> On Sun, Apr 15, 2012 at 12:48 AM, ocset <ocset at the800group.com> wrote:
>>
>>>  Thanks Nandy
>>>
>>> I am not using the GSM device in this test and have tried changing the
>>> disconnect signalling without any luck. When I hang up the phones, nothing
>>> is recorder in the Freeswitch log (log level 7). Is there a way to trap the
>>> disconnect signal by writing a "lua" script or would that info already be
>>> written to the log if it was being received?.
>>>
>>> What I don't understand is that when I use the SIP gateway for the
>>> outgoing leg of the call, then the disconnect signal from the two phones is
>>> received. But when I use the PSTN line as the outgoing leg, the disconnect
>>> signal is not received from either phone?
>>>
>>> Anyone got any suggestions how I would debug this?
>>>
>>> Thanks again for your help.
>>> O
>>>
>>>
>>>
>>> On 04/13/2012 03:40 PM, Nandy Dagondon wrote:
>>>
>>> check the line supervision signalling of your GSM and the GXW4104. there
>>> are several disconnect signalling used: current disconnect, polarity
>>> reversal or busy tones. your FXO port must be set accordingly.
>>>
>>>  On Fri, Apr 13, 2012 at 3:09 PM, ocset <ocset at the800group.com> wrote:
>>>
>>>> Hi
>>>>
>>>> I have successfully managed to get FreeSwitch to bridge an incoming call
>>>> on a PSTN line to an external mobile phone using a second PSTN line with
>>>> the help of a GXW4104 FSO gateway.
>>>>
>>>> I am now having a serious issue where the call never gets terminated
>>>> when the parties hang up. The only way to end the call is to either
>>>> shut-down FreeSwitch or turn off the GXW4104. If I bridge the call to a
>>>> SIP gateway (PennyTel) instead of the second PSTN line, then it works as
>>>> expected when both parties hang up.
>>>>
>>>> Here is what I have tested
>>>>
>>>> 1. Call -> PSTN line1 (GXW4140) -> PennyTel SIP gateway -> external
>>>> mobile phone -- call end when both parties hang up.
>>>> 2. Call -> PSTN line1 (GXW4104) ->  PSTN line2 (GXW4104) -> external
>>>> mobile phone -- call never end, even when both parties hang up.
>>>>
>>>> Is there a parameter I should be setting to help solve this issue or
>>>> some code that I should implement to test for this situation?
>>>>
>>>> I have also noticed that with test scenario 1 (PSTN line + SIP Gateway),
>>>> if the person being called does not end the call on their side, then
>>>> that leg of the bridge remains active.
>>>>
>>>> ps. The second PSTN line is not a real copper line. It is a PennyTel
>>>> VOIP line that is managed by the modem and connected to the GXW4104 via
>>>> phone cable. So, as far as the GXW4104 is concerned, it is just another
>>>> POTS line. I don't have a second PSTN phone line so can't test any other
>>>> way.
>>>>
>>>> Thanks in advance
>>>> O.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jean.marc.hyppolite at gmail.com  Mon Apr 16 03:02:42 2012
From: jean.marc.hyppolite at gmail.com (Jean-Marc Hyppolite)
Date: Sun, 15 Apr 2012 19:02:42 -0400
Subject: [Freeswitch-users] Help with tone detection
Message-ID: <4f8b539b.c83ae00a.1adc.fffffb33@mx.google.com>

Hello everyone,

 

Is it possible to have some help in setting up mod_spandsp for call progress
tone detection? I have not been successful thus far. I keep getting an
"error reading frame" message like this one below.

 

============================================================================
===================

2012-04-09 16:03:22.550303 [DEBUG] mod_spandsp_dsp.c:379
(sofia/internal/sip:1000 at 184.162.247.104:32986) Starting tone detection for
'1'

2012-04-09 16:03:22.550303 [INFO] mod_spandsp_dsp.c:411
(sofia/internal/sip:1000 at 184.162.247.104:32986) initializing tone detector

2012-04-09 16:03:22.550303 [DEBUG] switch_core_media_bug.c:462 Attaching BUG
to sofia/internal/sip:1000 at 184.162.247.104:32986

2012-04-09 16:03:22.590301 [DEBUG] switch_core_io.c:340 Setting BUG Codec
PCMU:0

2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:422
(sofia/internal/sip:1000 at 184.162.247.104:32986) error reading frame

2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:447
(sofia/internal/sip:1000 at 184.162.247.104:32986) destroying tone detector

============================================================================
===================

 

My spandsp.conf.xml file:

 

============================================================================
===================

<configuration name="spandsp.conf" description="Tone detector descriptors">

   <descriptors>

     <!-- These tones are defined in Annex to ITU Operational Bulletin No.
781 - 1.II.2003 -->

     <!-- Various Tones Used in National Networks (According to ITU-T
Recommendation E.180)(03/1998) -->

 

     <!-- North America -->

     <descriptor name="1">

       <tone name="CED_TONE">

         <element freq1="2100" freq2="0" min="500" max="0"/>

       </tone>

       <tone name="SIT">

         <element freq1="950" freq2="0" min="256" max="400"/>

         <element freq1="1400" freq2="0" min="256" max="400"/>

         <element freq1="1800" freq2="0" min="256" max="400"/>

       </tone>

       <tone name="REORDER_TONE">

         <element freq1="480" freq2="620" min="224" max="272"/>

         <element freq1="0" freq2="0" min="224" max="272"/>

       </tone>

       <tone name="BUSY_TONE">

         <element freq1="480" freq2="620" min="464" max="516"/>

         <element freq1="0" freq2="0" min="464" max="516"/>

       </tone>

     </descriptor>

   </descriptors>

</configuration>

============================================================================
===================

 

Thank you for your help.

 

Jean-Marc.

 

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From msc at freeswitch.org  Mon Apr 16 07:15:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 15 Apr 2012 20:15:10 -0700
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
	<CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
	<CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>
Message-ID: <CAKzWOxUhS0Cd_use8jgkJwi=z6_r8iOJattf43SN-_98fe+cUQ@mail.gmail.com>

On Fri, Apr 13, 2012 at 4:54 PM, Wesley Akio <wesleyakio at tuntscorp.com>wrote:

> Please let me know if you figure it out! I only have ISDN on my production
> boxes and sip providers over here are not really into T.38 therefore it's a
> bit complicated to debug...
>
>
FYI, there's also "execute_on_fax_result" which may be your best choice as
it lets execute something regardless of success or failure. Apologies but
it looks like this chan var never made it to the wiki. If anyone is
available to add it I would be most appreciative.

-MC
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From gcd at i.ph  Mon Apr 16 10:25:14 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 16 Apr 2012 14:25:14 +0800
Subject: [Freeswitch-users] Authenticate external calls problem
In-Reply-To: <C06636AC-6827-4AC2-B8E1-CEDD6848D290@me.com>
References: <C06636AC-6827-4AC2-B8E1-CEDD6848D290@me.com>
Message-ID: <CAHKqMb7+BeuYf9hYtVWNeorn4Npvfa5PS88eNYu2xEEjhSafCA@mail.gmail.com>

i think you should define Netelip as sofia_profiles/external/netelip.xml
because you should register to Netelip. you define netelip in
conf/directory/default/example.conf.xml when you require netelip to
register to your FS.

On Sat, Apr 14, 2012 at 9:11 AM, Jose Miguel Sucasas Mejuto <
jmsucasas at me.com> wrote:

> Hello,
>
>    I have installed freeSwitch 1.1beta from the git repository
> (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000).  My
> SIP provider (Netelip) is defined in
> conf/directory/default/example.conf.xml:
>
> <include>
>   <user id="$${default_provider}">
>     <gateways>
>       <gateway name="$${default_provider}">
> <param name="username" value="$${default_provider_username}"/>
> <param name="password" value="$${default_provider_password}"/>
> <param name="from-user" value="$${default_provider_username}"/>
> <param name="from-domain" value="$${default_provider_from_domain}"/>
> <param name="expire-seconds" value="600"/>
> <param name="register" value="$${default_provider_register}"/>
> <param name="retry-seconds" value="30"/>
> <param name="extension" value="$${default_provider_contact}"/>
> <param name="auth-calls" value="false"/>
> <!--<param name="contact-params" value="domain_name=$${domain}"/>-->
> <param name="context" value="public"/>
>       </gateway>
>     </gateways>
>     <params>
>       <param name="password" value="$${default_provider_password}"/>
>     </params>
>   </user>
> </include>
>
> and doesn't authenticate inbound calls (insecure=invite in asterisk
> language). When I receive an inbound call,  I receive the following error:
>
> freeswitch at internal>
> recv 1000 bytes from udp/[194.140.135.80]:5060 at 23:23:10.163377:
>    ------------------------------------------------------------------------
>    INVITE sip:34810101189 at sukmad.dyndns-home.com SIP/2.0
>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport
>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>    To: <sip:34810101189 at sukmad.dyndns-home.com>
>    Contact: <sip:916495242 at 194.140.135.80>
>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>    CSeq: 102 INVITE
>    User-Agent: Netelip
>    Max-Forwards: 70
>    Date: Fri, 13 Apr 2012 23:23:51 GMT
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>    Supported: replaces
>    Content-Type: application/sdp
>    Content-Length: 425
>
>    v=0
>    o=root 1249 1249 IN IP4 194.140.135.80
>    s=session
>    c=IN IP4 194.140.135.80
>    t=0 0
>    m=audio 16474 RTP/AVP 8 0 97 3 18 4 101
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:97 iLBC/8000
>    a=fmtp:97 mode=30
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:4 G723/8000
>    a=fmtp:4 annexa=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=silenceSupp:off - - - -
>    a=ptime:20
>    a=sendrecv
>    ------------------------------------------------------------------------
> send 382 bytes to udp/[194.140.135.80]:5060 at 23:23:10.164409:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>    To: <sip:34810101189 at sukmad.dyndns-home.com>
>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>    CSeq: 102 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31
> 23-42-27 +0000
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2012-04-14 01:23:10.162331 [DEBUG] sofia.c:7598 IP 194.140.135.80 Rejected
> by acl "domains". Falling back to Digest auth.
> send 871 bytes to udp/[194.140.135.80]:5060 at 23:23:10.166026:
>    ------------------------------------------------------------------------
> *   SIP/2.0 407 Proxy Authentication Required*
>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>    To: <sip:34810101189 at sukmad.dyndns-home.com>;tag=SaBHcvmm7evNF
>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>    CSeq: 102 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31
> 23-42-27 +0000
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Proxy-Authenticate: Digest realm="194.140.135.80",
> nonce="bc1571d4-0957-4609-a887-7c3359b9ae9c", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
> I have read in forums that the solution to this problem is to add the IP
> to the acl.conf.xml, like this:
>
>     <list name="domains" default="deny">
>       <!-- domain= is special it scans the domain from the directory to
> build the ACL -->
>       <node type="allow" domain="$${domain}"/>
>       <!-- use cidr= if you wish to allow ip ranges to this domains acl.
> -->
>       *<node type="allow" cidr="194.140.135.80/24"/> *
>     </list>
>
> But my SIP provider have a dynamic IP, and it's behind a NAT.  If my SIP
> provider will use any IP, ?Where can I do this without enabling all IPs to
> enter calls in my system?
>
>
> Thanks in advance,
>
> Regards.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gcd at i.ph  Mon Apr 16 10:26:07 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Mon, 16 Apr 2012 14:26:07 +0800
Subject: [Freeswitch-users] Authenticate external calls problem
In-Reply-To: <CAHKqMb7+BeuYf9hYtVWNeorn4Npvfa5PS88eNYu2xEEjhSafCA@mail.gmail.com>
References: <C06636AC-6827-4AC2-B8E1-CEDD6848D290@me.com>
	<CAHKqMb7+BeuYf9hYtVWNeorn4Npvfa5PS88eNYu2xEEjhSafCA@mail.gmail.com>
Message-ID: <CAHKqMb4LeDcq8vUwmuWTe0AF6hXK1=C-0YbR0_U3oFtzJgi-0A@mail.gmail.com>

correction ... it should be in conf/sip_profiles/external/netelip.xml. :-)

On Mon, Apr 16, 2012 at 2:25 PM, Nandy Dagondon <gcd at i.ph> wrote:

> i think you should define Netelip as sofia_profiles/external/netelip.xml
> because you should register to Netelip. you define netelip in
> conf/directory/default/example.conf.xml when you require netelip to
> register to your FS.
>
> On Sat, Apr 14, 2012 at 9:11 AM, Jose Miguel Sucasas Mejuto <
> jmsucasas at me.com> wrote:
>
>> Hello,
>>
>>    I have installed freeSwitch 1.1beta from the git repository
>> (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000).  My
>> SIP provider (Netelip) is defined in
>> conf/directory/default/example.conf.xml:
>>
>> <include>
>>   <user id="$${default_provider}">
>>     <gateways>
>>       <gateway name="$${default_provider}">
>> <param name="username" value="$${default_provider_username}"/>
>>  <param name="password" value="$${default_provider_password}"/>
>> <param name="from-user" value="$${default_provider_username}"/>
>>  <param name="from-domain" value="$${default_provider_from_domain}"/>
>> <param name="expire-seconds" value="600"/>
>>  <param name="register" value="$${default_provider_register}"/>
>> <param name="retry-seconds" value="30"/>
>>  <param name="extension" value="$${default_provider_contact}"/>
>> <param name="auth-calls" value="false"/>
>>  <!--<param name="contact-params" value="domain_name=$${domain}"/>-->
>> <param name="context" value="public"/>
>>       </gateway>
>>     </gateways>
>>     <params>
>>       <param name="password" value="$${default_provider_password}"/>
>>     </params>
>>   </user>
>> </include>
>>
>> and doesn't authenticate inbound calls (insecure=invite in asterisk
>> language). When I receive an inbound call,  I receive the following error:
>>
>> freeswitch at internal>
>> recv 1000 bytes from udp/[194.140.135.80]:5060 at 23:23:10.163377:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:34810101189 at sukmad.dyndns-home.com SIP/2.0
>>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport
>>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>>    To: <sip:34810101189 at sukmad.dyndns-home.com>
>>    Contact: <sip:916495242 at 194.140.135.80>
>>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>>    CSeq: 102 INVITE
>>    User-Agent: Netelip
>>    Max-Forwards: 70
>>    Date: Fri, 13 Apr 2012 23:23:51 GMT
>>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
>> INFO
>>    Supported: replaces
>>    Content-Type: application/sdp
>>    Content-Length: 425
>>
>>    v=0
>>    o=root 1249 1249 IN IP4 194.140.135.80
>>    s=session
>>    c=IN IP4 194.140.135.80
>>    t=0 0
>>    m=audio 16474 RTP/AVP 8 0 97 3 18 4 101
>>    a=rtpmap:8 PCMA/8000
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:97 iLBC/8000
>>    a=fmtp:97 mode=30
>>    a=rtpmap:3 GSM/8000
>>    a=rtpmap:18 G729/8000
>>    a=fmtp:18 annexb=no
>>    a=rtpmap:4 G723/8000
>>    a=fmtp:4 annexa=no
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=silenceSupp:off - - - -
>>    a=ptime:20
>>    a=sendrecv
>>
>>  ------------------------------------------------------------------------
>> send 382 bytes to udp/[194.140.135.80]:5060 at 23:23:10.164409:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
>>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>>    To: <sip:34810101189 at sukmad.dyndns-home.com>
>>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>>    CSeq: 102 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31
>> 23-42-27 +0000
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> 2012-04-14 01:23:10.162331 [DEBUG] sofia.c:7598 IP 194.140.135.80
>> Rejected by acl "domains". Falling back to Digest auth.
>> send 871 bytes to udp/[194.140.135.80]:5060 at 23:23:10.166026:
>>
>>  ------------------------------------------------------------------------
>> *   SIP/2.0 407 Proxy Authentication Required*
>>    Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060
>>    From: "916495242" <sip:916495242 at 194.140.135.80>;tag=as7a2a97e5
>>    To: <sip:34810101189 at sukmad.dyndns-home.com>;tag=SaBHcvmm7evNF
>>    Call-ID: 6c47a4064ef8d0641da415d2121e3226 at 194.140.135.80
>>    CSeq: 102 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31
>> 23-42-27 +0000
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
>> sla, include-session-description, presence.winfo, message-summary, refer
>>    Proxy-Authenticate: Digest realm="194.140.135.80",
>> nonce="bc1571d4-0957-4609-a887-7c3359b9ae9c", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>> I have read in forums that the solution to this problem is to add the IP
>> to the acl.conf.xml, like this:
>>
>>     <list name="domains" default="deny">
>>       <!-- domain= is special it scans the domain from the directory to
>> build the ACL -->
>>       <node type="allow" domain="$${domain}"/>
>>       <!-- use cidr= if you wish to allow ip ranges to this domains acl.
>> -->
>>       *<node type="allow" cidr="194.140.135.80/24"/> *
>>     </list>
>>
>> But my SIP provider have a dynamic IP, and it's behind a NAT.  If my SIP
>> provider will use any IP, ?Where can I do this without enabling all IPs to
>> enter calls in my system?
>>
>>
>> Thanks in advance,
>>
>> Regards.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From william.king at quentustech.com  Mon Apr 16 10:35:14 2012
From: william.king at quentustech.com (William King)
Date: Sun, 15 Apr 2012 23:35:14 -0700
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <CAKzWOxUhS0Cd_use8jgkJwi=z6_r8iOJattf43SN-_98fe+cUQ@mail.gmail.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
	<CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
	<CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>
	<CAKzWOxUhS0Cd_use8jgkJwi=z6_r8iOJattf43SN-_98fe+cUQ@mail.gmail.com>
Message-ID: <4F8BBDA2.2030002@quentustech.com>

Out of curiosity and for future documentation, do you know which channel
variable has the fax result status?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/15/2012 08:15 PM, Michael Collins wrote:
>
> On Fri, Apr 13, 2012 at 4:54 PM, Wesley Akio <wesleyakio at tuntscorp.com
> <mailto:wesleyakio at tuntscorp.com>> wrote:
>
>     Please let me know if you figure it out! I only have ISDN on my
>     production boxes and sip providers over here are not really into
>     T.38 therefore it's a bit complicated to debug...
>
>
> FYI, there's also "execute_on_fax_result" which may be your best
> choice as it lets execute something regardless of success or failure.
> Apologies but it looks like this chan var never made it to the wiki.
> If anyone is available to add it I would be most appreciative.
>
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From azza.miled at gmail.com  Mon Apr 16 14:21:22 2012
From: azza.miled at gmail.com (azza miled)
Date: Mon, 16 Apr 2012 11:21:22 +0100
Subject: [Freeswitch-users] FreeSWITCH and Opensips
Message-ID: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>

Hi!
I want that my FeeSWITCH receives calls from my opensips' users without
requiring authentication. In other words, I want that my FS receives
unauthenticated calls, only from the IP address of my OpenSIPS server. any
help, please???
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From khorsmann at gmail.com  Mon Apr 16 14:32:29 2012
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 16 Apr 2012 12:32:29 +0200
Subject: [Freeswitch-users] FreeSWITCH and Opensips
In-Reply-To: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
References: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
Message-ID: <CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>

Hi.

You can use acl for that.

<http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.html>

2012/4/16 azza miled <azza.miled at gmail.com>:
> Hi!
> I want that my FeeSWITCH receives calls from my opensips' users without
> requiring authentication. In other words, I want that my FS receives
> unauthenticated calls, only from the IP address of my OpenSIPS server. any
> help, please???



-- 
Mit freundlichen Gr??en
*Karsten Horsmann*


From cmrienzo at gmail.com  Mon Apr 16 16:28:52 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 16 Apr 2012 08:28:52 -0400
Subject: [Freeswitch-users] Nuance mod_unimrcp ASR
In-Reply-To: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
References: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
Message-ID: <CAD-qRkg-ZYXRcSpyg4ORkZLuVb_kOXdUhrFKWWeOGnteMOEfYQ@mail.gmail.com>

1. Use the dialplan apps play_and_detect_speech/detect_speech over event
socket or lua/javascript
2. mod_httapi has an interface to speech recognition - I've never tried this

Regardless of interface used, you have to write an app to parse the NLSML
result from the server.



On Sat, Apr 14, 2012 at 3:06 PM, Anita Hall <anita.hall at simmortel.com>wrote:

> Hi
>
> I am making a FreeSWITCH application for ASR using Nuance Speech Server
> 5.0 and mod_unimrcp. What are my options ?
>
> regards,
> Anita
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Hector.Geraldino at ipsoft.com  Mon Apr 16 17:40:05 2012
From: Hector.Geraldino at ipsoft.com (Hector Geraldino)
Date: Mon, 16 Apr 2012 09:40:05 -0400
Subject: [Freeswitch-users] Nuance mod_unimrcp ASR
In-Reply-To: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
References: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD022E872F10@NY1-EXMB-01.ip-soft.net>

I have a setup similar to yours, and I'm using mod_event_socket with detect_speech command/DETECTED_SPEECH event to get the transcription.

Although I think the product name is Nuance Recognizer (NR9), the Speech Server covers only the MRCP protocol and relies on NR9 for the speech recognition.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anita Hall
Sent: Saturday, April 14, 2012 3:07 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Nuance mod_unimrcp ASR

Hi

I am making a FreeSWITCH application for ASR using Nuance Speech Server 5.0 and mod_unimrcp. What are my options ?

regards,
Anita
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From tech at fahrwerk-berlin.de  Mon Apr 16 17:50:50 2012
From: tech at fahrwerk-berlin.de (Fahrwerk Kurierkollektiv (Technik))
Date: Mon, 16 Apr 2012 15:50:50 +0200
Subject: [Freeswitch-users] mod_callcenter - using moh and ring tones
 when phone is ringing; possible?
In-Reply-To: <4F84B278.3070003@fahrwerk-berlin.de>
References: <4F830CB0.6020301@fahrwerk-berlin.de>	<CAGKPtBsuhEvmbdo7W-mfdB8pOK58tL3WGCS232xQjz2WM8pjxw@mail.gmail.com>	<CAC-LwPMqFPudn8OT8fby1RUOnyr83YV9je6oPcdFsJUna=3zHg@mail.gmail.com>
	<4F84B278.3070003@fahrwerk-berlin.de>
Message-ID: <4F8C23BA.3020404@fahrwerk-berlin.de>

Hi Vik,

>> On 10.04.2012 16:17, Vik Killa wrote:
>>  Have you tried setting the variable 'cc_moh_override' ?
>>  I believe you can set it like
>>    <action application="set"
>>  data="cc_moh_override=tone_stream://%(400,200,400,450);%(400,2200,400,450)"/>

I've tried this now. However, no success: Using this overwrites the moh, 
and gives me only ringtones.

Anyone else?

Thanks,
Georg


From mitch.capper at gmail.com  Mon Apr 16 17:57:00 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 16 Apr 2012 06:57:00 -0700
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F8BBDA2.2030002@quentustech.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
	<CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
	<CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>
	<CAKzWOxUhS0Cd_use8jgkJwi=z6_r8iOJattf43SN-_98fe+cUQ@mail.gmail.com>
	<4F8BBDA2.2030002@quentustech.com>
Message-ID: <CALn233rNJyFvjrrSH2q57F360_gQ=U1NEuCy+q6ZKoQqMpA_Tw@mail.gmail.com>

I use variable_fax_success / variable_fax_result_text / variable_fax_result_code
for when faxes are done.

~Mitch


From sat at calgaryit.com  Mon Apr 16 18:02:58 2012
From: sat at calgaryit.com (George Sapak)
Date: Mon, 16 Apr 2012 08:02:58 -0600 (MDT)
Subject: [Freeswitch-users] Voicemail hangups
In-Reply-To: <1422889126.160382.1334584443659.JavaMail.root@server3>
Message-ID: <79894403.160416.1334584978542.JavaMail.root@server3>

I have a quite a few people that want to leave voice mails and they say they are being hung upon as if they were not recording a message, I think its usually from people that are quitter speakers, these are my settings in my voicemail file:

<param name="record-silence-threshold" value="1"/>
<param name="record-silence-hits" value="999"/>

is there a way to set the params so the vm records event if there is almost total silence

Thank you, George.


From azza.miled at gmail.com  Mon Apr 16 19:35:51 2012
From: azza.miled at gmail.com (azza miled)
Date: Mon, 16 Apr 2012 16:35:51 +0100
Subject: [Freeswitch-users] Integrating FreeWITCH and Opensips
In-Reply-To: <CAKV-jFxGLBybV9tbXmwvKGYMCQECHOxXQEoAzxpyKK1iF8O97g@mail.gmail.com>
References: <CAGQdkzTgfKEZT5+vEvTFot0-XL0Y4adbWwK4kvzOOf_8Q5etOg@mail.gmail.com>
	<CAKV-jFxGLBybV9tbXmwvKGYMCQECHOxXQEoAzxpyKK1iF8O97g@mail.gmail.com>
Message-ID: <CAGQdkzQtvNHk9C6wkm5Te+V8cfQFP6wfwCFYkEpw+BS554mO6Q@mail.gmail.com>

Thanks for the response, but evec with these steps, I still have the error
"call failed:proxy authentication required" when I dial the IVR extension
from an opensips' phone. Do any one know, what shall I do???

2012/4/15 Brian Foster <bdfoster at endigotech.com>

> There are tutorials on how to make an IVR on the wiki, see
> http://wiki.freeswitch.org and search for IVR.
> On Apr 15, 2012 6:27 PM, "azza miled" <azza.miled at gmail.com> wrote:
>
>> Hi!
>> I want to use FreeSWITCH to handle IVR for my opensips's users.I have no
>> idea where to start.A detailed tutoriel will be helpful.
>>  Can any one help me, please?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From azza.miled at gmail.com  Mon Apr 16 19:36:42 2012
From: azza.miled at gmail.com (azza miled)
Date: Mon, 16 Apr 2012 16:36:42 +0100
Subject: [Freeswitch-users] FreeSWITCH and Opensips
In-Reply-To: <CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>
References: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
	<CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>
Message-ID: <CAGQdkzRfjkbiaWuqTpC-3aHv2A-KhrLRkQ_Bw3=+Vq+9qZPf3w@mail.gmail.com>

Thanks for the response, but evec with these steps, I still have the error
"call failed:proxy authentication required" when I dial the IVR extension
from an opensips' phone. Do any one know, what shall I do???

2012/4/16 Karsten Horsmann <khorsmann at gmail.com>

> Hi.
>
> You can use acl for that.
>
> <
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-April/003067.html
> >
>
> 2012/4/16 azza miled <azza.miled at gmail.com>:
> > Hi!
> > I want that my FeeSWITCH receives calls from my opensips' users without
> > requiring authentication. In other words, I want that my FS receives
> > unauthenticated calls, only from the IP address of my OpenSIPS server.
> any
> > help, please???
>
>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Mon Apr 16 19:41:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 08:41:21 -0700
Subject: [Freeswitch-users] Faxing via FreeSWITCH documentation
In-Reply-To: <4F8BBDA2.2030002@quentustech.com>
References: <4F88642F.9010706@gmail.com>
	<CAKV-jFxtABgFHV4ahvuuu8ncg+rH6R+xRJzYMC8gRKy6S+31Rw@mail.gmail.com>
	<4F886BC0.2010306@gmail.com>
	<CAKV-jFwSz-JEKRQBEqNSRfcTiQi+k_ebRjfUMifoz_4_vZaOvw@mail.gmail.com>
	<CAGTsDSuux78EYGs40DPu=LtvqzK=dj+1BdZPdmXJQAAujTq33w@mail.gmail.com>
	<CAKV-jFzSJ8uB2o9BvSuXABX0wfpw-UFF-GoTD87h09D1xpUBNA@mail.gmail.com>
	<CAGTsDSvtAZmt5cpuLr7roOYmxEvdz-uXNr7ak=qhR10PVHd_dQ@mail.gmail.com>
	<CAKzWOxUhS0Cd_use8jgkJwi=z6_r8iOJattf43SN-_98fe+cUQ@mail.gmail.com>
	<4F8BBDA2.2030002@quentustech.com>
Message-ID: <CAKzWOxUDvbQG-FvcP4Hea=E+g_GNjfP++sU4dxzwmp4xiUVf2A@mail.gmail.com>

On Sun, Apr 15, 2012 at 11:35 PM, William King <william.king at quentustech.com
> wrote:

>  Out of curiosity and for future documentation, do you know which channel
> variable has the fax result status?
>
According to the source, it's "fax_result_code"
-MC
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From msc at freeswitch.org  Mon Apr 16 20:17:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 09:17:22 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call: DNS Best
	Practices in a VoIP Environment
Message-ID: <CAKzWOxWdy6ZhgD5yVPY7uOdbYpq3ca_xFEBw=eUnUGNiYCL9Hg@mail.gmail.com>

Greetings,

We'd like to have everyone join us on Wednesday, April 18th for a great
presentation to be given by our very own Travis Cross. He will be
discussing DNS Best Practices in a VoIP Environment. I'm definitely looking
forward to this discussion as this is an important topic and formal
documentation isn't too widely dispensed at the moment. I will be taking
notes on this discussion and creating a corresponding wiki page. The agenda
page is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_18

To make this discussion more meaningful I would like everyone to be willing
to share their personal experiences and knowledge, as well as any questions
that they have on the subject. We have a number of community members whose
skill we can draw upon, and the more questions we ask, the more information
we can collect and put into formal documentation.

Talk to you this Wednesday!
-Michael
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From bdfoster at endigotech.com  Mon Apr 16 20:47:26 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 16 Apr 2012 12:47:26 -0400
Subject: [Freeswitch-users] Integrating FreeWITCH and Opensips
In-Reply-To: <CAGQdkzQtvNHk9C6wkm5Te+V8cfQFP6wfwCFYkEpw+BS554mO6Q@mail.gmail.com>
References: <CAGQdkzTgfKEZT5+vEvTFot0-XL0Y4adbWwK4kvzOOf_8Q5etOg@mail.gmail.com>
	<CAKV-jFxGLBybV9tbXmwvKGYMCQECHOxXQEoAzxpyKK1iF8O97g@mail.gmail.com>
	<CAGQdkzQtvNHk9C6wkm5Te+V8cfQFP6wfwCFYkEpw+BS554mO6Q@mail.gmail.com>
Message-ID: <CAKV-jFyHw85ddrn9f5wasyqdcR7R8jGTcujffoZVqPNHEuSjpA@mail.gmail.com>

Looks like you have another thread with this same issue open already, so
it's best to continue the discussion there.

***Message Thread Closed***

-BDF

On Mon, Apr 16, 2012 at 11:35 AM, azza miled <azza.miled at gmail.com> wrote:

> Thanks for the response, but evec with these steps, I still have the error
> "call failed:proxy authentication required" when I dial the IVR extension
> from an opensips' phone. Do any one know, what shall I do???
>
>
> 2012/4/15 Brian Foster <bdfoster at endigotech.com>
>
>> There are tutorials on how to make an IVR on the wiki, see
>> http://wiki.freeswitch.org and search for IVR.
>> On Apr 15, 2012 6:27 PM, "azza miled" <azza.miled at gmail.com> wrote:
>>
>>> Hi!
>>> I want to use FreeSWITCH to handle IVR for my opensips's users.I have no
>>> idea where to start.A detailed tutoriel will be helpful.
>>>  Can any one help me, please?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
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The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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From jmoran at secureachsystems.com  Mon Apr 16 23:01:14 2012
From: jmoran at secureachsystems.com (Jason Moran)
Date: Mon, 16 Apr 2012 15:01:14 -0400
Subject: [Freeswitch-users] How to update a live system with very little
	downtime (How does 'make current' work?)
Message-ID: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>

Sorry for such a basic question:

1.       If I do a 'make current' while a system is running it will not
have any effect until I restart FS, correct?

2.       What if 'make current' fails? What if it fails and the server
gets restarted, will FS work?

3.       Is it more advisable to stop FS, do the 'make current', then
start FS even though it will have a longer down time?

 

I know it's best to keep FS up-to-date - but it is difficult when you
only have 1 system available and it is expected to have 100% uptime..

Thanks,
Jason

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From freeswitch at earthspike.net  Tue Apr 17 00:23:09 2012
From: freeswitch at earthspike.net (John)
Date: Mon, 16 Apr 2012 21:23:09 +0100
Subject: [Freeswitch-users] How to update a live system with very little
 downtime (How does 'make current' work?)
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>
Message-ID: <4F8C7FAD.9070509@earthspike.net>

Jason,

This may not be the best way but is the way I do it on my Ubuntu
10.04-based system.  I found that make current does the build and
install without pausing.  On my li'l Atom-based system, the build can
take a while, meaning that the install could be at an
unpredictable/inconvenient time.  Also, I have a Sangoma ISDN card, so
the wanpipe kernel drivers need rebuilding on every kernel update; I
take this opportunity to do everything else, like this:

0. Backup all your config files, recordings, logs, etc.

1. Download any updated Sangoma drivers (libsngisdn and wanpipe). 
Install the libsngisdn drivers. On my system, they also need to be
copied from /lib64 to /usr/lib64 and the various symlinks remade before
rerunning ldconfig. None of this affects the running system as the
libsngisdn library version is hard coded into the executables at FreeTDM
build time.  Decompress the wanpipe drivers but don't build them (this
should be done after the kernel upgrade).

2. In /usr/local/src/freeswitch (YMMV) do a git pull.  I keep an eye on
the mailing list so know whether I need to do a ./bootstrap.sh or not. 
If in doubt, do.  Then `./configure` and `make` as normal.  (Don't do
`make install` yet.)

You then have 2 options: either do the kernel upgrade and rebuild the
wanpipe drivers with the current FreeSWITCH, then upgrade FreeSWITCH, or
do FreeSWITCH first then the kernel upgrade.  Doing both at once is mad
as if something goes wrong you then have a problem identifying what
caused it.  Assuming the first approach (my preferred as it is easier
(for me) to roll the kernel back).

3. Do the kernel upgrade (in my case, `apt-get dist-upgrade`,
`reboot`).  mod-freetdm will fail to load as wanpipe will be to the
wrong kernel version; this is expected. This will cause the first outage.

4. Upgrade the wanpipe drivers. If the wanpipe drivers have been
upgraded themselves, then you need to `./Setup install` otherwise just
`./Setup drivers`.  Use the same options as when first building them.
`wanrouter start` to get the wanpipe devices live again.

5. At the fs_cli prompt, `load mod_freetdm` should now succeed.  You may
wish to reboot the system again to check that all comes up from a cold
start as it should do. Here ends the first outage (usually about 10
minutes), but you could run it straight into the second if you wish, or
pause here if you need to.

6. Now comes the time to upgrade FreeSWITCH. This will cause an outage.
Stop FreeSWITCH then `make install`.  Start FreeSWITCH; all should be
well.  This normally only takes about 5-10mins on my system.

This doesn't answer your Y question but might answer your X question.
[http://www.perlmonks.org/?node_id=542341]

John


On 16/04/12 20:01, Jason Moran wrote:
>
> Sorry for such a basic question:
>
> 1.       If I do a 'make current' while a system is running it will
> not have any effect until I restart FS, correct?
>
> 2.       What if 'make current' fails? What if it fails and the server
> gets restarted, will FS work?
>
> 3.       Is it more advisable to stop FS, do the 'make current', then
> start FS even though it will have a longer down time?
>
>  
>
> I know it's best to keep FS up-to-date -- but it is difficult when you
> only have 1 system available and it is expected to have 100% uptime..
>
> Thanks,
> Jason
>
>

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From msc at freeswitch.org  Tue Apr 17 00:52:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 13:52:08 -0700
Subject: [Freeswitch-users] FreeSWITCH and Opensips
In-Reply-To: <CAGQdkzRfjkbiaWuqTpC-3aHv2A-KhrLRkQ_Bw3=+Vq+9qZPf3w@mail.gmail.com>
References: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
	<CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>
	<CAGQdkzRfjkbiaWuqTpC-3aHv2A-KhrLRkQ_Bw3=+Vq+9qZPf3w@mail.gmail.com>
Message-ID: <CAKzWOxUJqd+jHaWbjhCAVtm6XaYS3XzRtnZ1b3mf97-GqNJVMA@mail.gmail.com>

On Mon, Apr 16, 2012 at 8:36 AM, azza miled <azza.miled at gmail.com> wrote:

> Thanks for the response, but evec with these steps, I still have the error
> "call failed:proxy authentication required" when I dial the IVR extension
> from an opensips' phone. Do any one know, what shall I do???
>
> Do you need SIP authentication between your FS box and your OpenSIPS box?
I'm guessing not. If that's the case then I recommend opening up this file:

/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml

Find the tag with this opening line:
<list name="domains" default="deny">

Add a new line in there like this:
<node type="allow" cidr="x.x.x.x/32"/>
where x.x.x.x is the IP address of your OpenSIPS server.

then go to fs_cli and issue this command: reloadacl

Lastly, you'll need to route the inbound calls from OpenSIPS through your
'public' dialplan context because that's where ACL-allowed inbound calls
normally get processed. You can either create your IVRs in the public
context or you can transfer the inbound call over to the default context
(or whatever context you are using.)

-MC
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From msc at freeswitch.org  Tue Apr 17 01:02:38 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 14:02:38 -0700
Subject: [Freeswitch-users] How to update a live system with very little
 downtime (How does 'make current' work?)
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>
Message-ID: <CAKzWOxUaAv3zSkmej1ansZMK2k_hAcM+J-OLy_8t=6Wj51hgkw@mail.gmail.com>

On Mon, Apr 16, 2012 at 12:01 PM, Jason Moran
<jmoran at secureachsystems.com>wrote:

>  Sorry for such a basic question:****
>
> **1.       **If I do a ?make current? while a system is running it will
> not have any effect until I restart FS, correct?****
>
> **2.       **What if ?make current? fails? What if it fails and the
> server gets restarted, will FS work?****
>
> **3.       **Is it more advisable to stop FS, do the ?make current?, then
> start FS even though it will have a longer down time?****
>
> ** **
>
> I know it?s best to keep FS up-to-date ? but it is difficult when you only
> have 1 system available and it is expected to have 100% uptime..****
>
> Thanks,
> Jason
>

Since you don't have a spare system (which is totally naughty, btw,
especially if your corporate overlords expect 100% uptime) then my next
question would be this: how are you set on CPU power, RAM, and disk space?
The reason I ask is that it is entirely possible to create a secondary git
clone of the FS repo and use a completely different install path when
running the configure script. You could "make current" in that secondary
repo and get a sort of preview of what would happen w/o touching your
production build env or install dirs. If the update goes well in your test
scenario then you could do the update in your production environment.

 It's kinda hackish but if you have spare disk space then it's a really
cheap option. It isn't perfect, but it does give you at least some
semblance of "try before you buy."

-MC
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From msc at freeswitch.org  Tue Apr 17 01:12:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 14:12:28 -0700
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05CF91@RSV16.realitysolutions.local>
References: <8C26A4FDAE599041A13EB499117D3C287CA27576@EX-MB-1.corp.atlasnetworks.us>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05CF91@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxXQ1LCp_22r+my8ZP894kY5v3Y0gf++hosctRKy4FyCXw@mail.gmail.com>

I still suspect something in FreeTDM. From Nathan's perspective that would
be "inside" FreeSWITCH since that's the TDM stack we use. I highly
recommend you open a trouble ticket with Sangoma and have each person who
is experiencing the issue put a comment on the case and offer to supply
configs, logs, and/or access that Sangoma would need to do troubleshooting.

-MC

On Fri, Apr 13, 2012 at 12:35 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

> That's very interesting, Nathan.
>
> Now that makes three of us, two using the Digium cards and one using a
> Sangoma.
>
>
> It very much looks like something FreeSwitch is doing, perhaps the DTMF
> detection is far too sensitive?  Are DTMF tones for an ISDN line inband or
> do they come in on the D-channel?
>
>
> Michael - any ideas what we can all do?
>
>
>
>
> From: Nathan Eisenberg [mailto:nathan at atlasnetworks.us]
> Sent: 13 April 2012 00:21
> To: Daniel Knaggs
> Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
>
> I'm having the exact issue you're describing, with the same card.
>  However, what's interesting to me is that if I fire up Asterisk instead of
> freeswitch, the phantom DTMF problem disappears entirely.
>
> So it's *something* inside Freeswitch, at least for me.  Would be
> interesting if you're able to do a similar test.
> Nathan Eisenberg
> Atlas Networks | Sr. Systems Administrator
> office: 206.577.3078  |  www.atlasnetworks.us
>  D
>
>
> Daniel Knaggs
> Software Developer
>
> Reality Solutions Ltd
> 1 Global Business Park
> Hamburg Road
> Kingston upon Hull
> East Yorkshire, HU7 0AE
>
> Tel:    01482 373104
> Mobile: 07932 408313
> Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
> http://
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 17 01:18:10 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 14:18:10 -0700
Subject: [Freeswitch-users] external sip profile - registered users get
 Originate Failed. Cause: USER_NOT_REGISTERED
In-Reply-To: <CAFArqsby4u24L4RbFjLN-N-y1aq9QUVpQyNAEJU_T5kk=gs-EA@mail.gmail.com>
References: <CAFArqsaUS_kQNNwL9LHRk9Ztq6vJ-b96W3qiHaQeGD==i3nGxA@mail.gmail.com>
	<CANBhyPAi0Ciu3ybpTX9caPGk_5ysK9OXKO=Sqky6Wo=XXgzmWw@mail.gmail.com>
	<CAFArqsby4u24L4RbFjLN-N-y1aq9QUVpQyNAEJU_T5kk=gs-EA@mail.gmail.com>
Message-ID: <CAKzWOxWSE82fLXpEiUdsUFMb1VRG3B4uxrCzQ_k4eBALPfeZ9Q@mail.gmail.com>

You can use any channel variable as well as the "special" variables listed
here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Built-In_Variables

Example:
  <condition field="network_addr" expression="^192\.168\.1\.1$"/>  <!--
network address=192.168.1.1 >

-MC

On Thu, Apr 12, 2012 at 3:51 AM, Karsten Horsmann <khorsmann at gmail.com>wrote:

> Hi Anton,
>
> because all clients using 5060 as default and i didnt want to inform 1000
> users
> that they must use an diffrent port.
>
> Also i use diffrent ip addresses for internal and external, so it
> should be clear
> where it comes from.
>
> So what can i do to reconnice FS the registered user from external?
>
>
> 2012/4/12 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> > You are using the same port 5060 for internal and external profile,
> that's
> > not good :) Why did you change default 5080 port for external profile?
>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 17 01:24:08 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 14:24:08 -0700
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CBAF0DC2.312AB%krice@freeswitch.org>
References: <CAE3mkNALxP5pVW9Ct3qb7KJy-k6HAhpD7vjnXGHAR1vgAUO+pA@mail.gmail.com>
	<CBAF0DC2.312AB%krice@freeswitch.org>
Message-ID: <CAKzWOxXFLQAFC0bYt=iCjFHoVVHs-YFOxMz9GUEGbP+9T7JGtQ@mail.gmail.com>

> Iirc it is sqlite3, but we have a defacto fork of SQLite running due to a
> variety of things we had to re-arrange in the code. Standard SQLite tools
> should still allow you to play with the .db files tho
>

FWIW, I've used the sqlite3 CLI program to query and manipulate the .db
files. I've never run into any drama or issues.
-MC
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From msc at freeswitch.org  Tue Apr 17 01:28:04 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 14:28:04 -0700
Subject: [Freeswitch-users] creating a conference for 2 separate parked
 UUIDs, please help
In-Reply-To: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
References: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
Message-ID: <CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>

Conferences get created on the fly. For example, in the default configs you
can call 3000 and it magically creates a conference named 3000-${domain} on
your system. For testing, try doing a uuid_transfer of these parked calls
and drop them into the dialplan that in turn sends them to the appropriate
conference:

uuid_transfer <uuid> 3000
or
uuid_transfer <uuid> 3000 XML default

Roll your own version of x3000 if you want the conference to be
unique/special/different.

-MC

On Sun, Apr 15, 2012 at 4:11 AM, Anton VG <anton.vazir at gmail.com> wrote:

> Guys,
>
> I need to create a conference for 2 UUID's , which already exists and
> parked. How do I create a new, empty conference and then send those 2
> UUIDs there?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Tue Apr 17 01:30:29 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 16 Apr 2012 14:30:29 -0700
Subject: [Freeswitch-users] Callgroup keeps ringing after answer - changed
	from shared extensions
Message-ID: <33B7D98B-6002-43CB-B8B7-DEFB30B34B2C@mgtech.com>

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. 

Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs.
Mario G

default.xml ring-group has (abbreviated to save space)------------------:
<group name="home">
	<users>
	  <user id="100" type="pointer"/>
	  <user id="101" type="pointer"/>
	..... more users
	</users>
      </group>
      
        <group name="work">
	<users>
	  <user id="200" type="pointer"/>
	  <user id="201" type="pointer"/>
	</users>
      </group>
    </groups>

Extension----------------------:
  <user id="101">
User:
    <params>
      <param name="password" value="$${default_password}"/>
    </params>
    <variables>
      <variable name="voicemail_domain" value="xxxxxx.com"/>
      <variable name="toll_allow" value="domestic,domestic10,international,local"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Front Desk"/>
      <variable name="effective_caller_id_number" value="101"/>
      <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
      <!-- callgroup needed to stop ringing on call pickup  -->
      <variable name="callgroup" value="home"/>
    </variables>
  </user>

Bridge command-----------------:
    <extension name="xxphone">
      <condition field="destination_number" expression="^2920$">
        <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
        <!--  Set the sayname variable with the callers name -->  
        <action application="execute_extension" data="2925 XML default"/>
        <!-- Next lines allow control after bridge timeout and hangup after bridge -->
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
	<action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>
        <!-- ring extensions only for ## seconds -->
        <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>
   
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From msc at freeswitch.org  Tue Apr 17 02:32:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 15:32:03 -0700
Subject: [Freeswitch-users] Help with tone detection
In-Reply-To: <4f8b539b.c83ae00a.1adc.fffffb33@mx.google.com>
References: <4f8b539b.c83ae00a.1adc.fffffb33@mx.google.com>
Message-ID: <CAKzWOxWSscZmAq4UM1CHVMhd3=0-GTwHgkSx0mM0pMVOd7cgcQ@mail.gmail.com>

I haven't been able to reproduce this issue. Can you post the dialplan that
you are using?
-MC

On Sun, Apr 15, 2012 at 4:02 PM, Jean-Marc Hyppolite <
jean.marc.hyppolite at gmail.com> wrote:

> Hello everyone,****
>
> ** **
>
> Is it possible to have some help in setting up mod_spandsp for call
> progress tone detection? I have not been successful thus far. I keep
> getting an ?error reading frame? message like this one below.****
>
> ** **
>
>
> ===============================================================================================
> ****
>
> 2012-04-09 16:03:22.550303 [DEBUG] mod_spandsp_dsp.c:379 (sofia/internal/
> sip:1000 at 184.162.247.104:32986) Starting tone detection for '1'****
>
> 2012-04-09 16:03:22.550303 [INFO] mod_spandsp_dsp.c:411 (sofia/internal/
> sip:1000 at 184.162.247.104:32986) initializing tone detector****
>
> 2012-04-09 16:03:22.550303 [DEBUG] switch_core_media_bug.c:462 Attaching
> BUG to sofia/internal/sip:1000 at 184.162.247.104:32986****
>
> 2012-04-09 16:03:22.590301 [DEBUG] switch_core_io.c:340 Setting BUG Codec
> PCMU:0****
>
> 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:422 (sofia/internal/
> sip:1000 at 184.162.247.104:32986) error reading frame****
>
> 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:447 (sofia/internal/
> sip:1000 at 184.162.247.104:32986) destroying tone detector****
>
>
> ===============================================================================================
> ****
>
> ** **
>
> My spandsp.conf.xml file:****
>
> ** **
>
>
> ===============================================================================================
> ****
>
> <configuration name=*"spandsp.conf"* description=*"Tone detector
> descriptors"*>****
>
>    <descriptors>****
>
>      <!-- These tones are defined in Annex to ITU Operational Bulletin
> No. 781 - 1.II.2003 -->****
>
>      <!-- Various Tones Used in National Networks (According to ITU-T
> Recommendation E.180)(03/1998) -->****
>
> ** **
>
>      <!-- North America -->****
>
>      <descriptor name=*"1"*>****
>
>        <tone name=*"CED_TONE"*>****
>
>          <element freq1=*"2100"* freq2=*"0"* min=*"500"* max=*"0"*/>****
>
>        </tone>****
>
>        <tone name=*"SIT"*>****
>
>          <element freq1=*"950"* freq2=*"0"* min=*"256"* max=*"400"*/>****
>
>          <element freq1=*"1400"* freq2=*"0"* min=*"256"* max=*"400"*/>****
>
>          <element freq1=*"1800"* freq2=*"0"* min=*"256"* max=*"400"*/>****
>
>        </tone>****
>
>        <tone name=*"REORDER_TONE"*>****
>
>          <element freq1=*"480"* freq2=*"620"* min=*"224"* max=*"272"*/>***
> *
>
>          <element freq1=*"0"* freq2=*"0"* min=*"224"* max=*"272"*/>****
>
>        </tone>****
>
>        <tone name=*"BUSY_TONE"*>****
>
>          <element freq1=*"480"* freq2=*"620"* min=*"464"* max=*"516"*/>***
> *
>
>          <element freq1=*"0"* freq2=*"0"* min=*"464"* max=*"516"*/>****
>
>        </tone>****
>
>      </descriptor>****
>
>    </descriptors>****
>
> </configuration>****
>
>
> ===============================================================================================
> ****
>
> ** **
>
> Thank you for your help.****
>
> ** **
>
> Jean-Marc.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 17 02:37:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 15:37:35 -0700
Subject: [Freeswitch-users] Nuance mod_unimrcp ASR
In-Reply-To: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
References: <CAGKPtBuOY79EH7Gb2OwiWASY8iLw4HE1DO75e-VnUPuxd6VtSg@mail.gmail.com>
Message-ID: <CAKzWOxV0cvWFEEdB3GYX92z4=wyTeoa_=PcFu7Aqn72wUyYpcg@mail.gmail.com>

You may also want to check out Vestec. They are probably a lot cheaper than
Nuance and will work every bit as well.
-MC

On Sat, Apr 14, 2012 at 12:06 PM, Anita Hall <anita.hall at simmortel.com>wrote:

> Hi
>
> I am making a FreeSWITCH application for ASR using Nuance Speech Server
> 5.0 and mod_unimrcp. What are my options ?
>
> regards,
> Anita
>
>
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From msc at freeswitch.org  Tue Apr 17 02:40:18 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Apr 2012 15:40:18 -0700
Subject: [Freeswitch-users] Voicemail hangups
In-Reply-To: <79894403.160416.1334584978542.JavaMail.root@server3>
References: <1422889126.160382.1334584443659.JavaMail.root@server3>
	<79894403.160416.1334584978542.JavaMail.root@server3>
Message-ID: <CAKzWOxXE=QWui8=h_zaXJQsMH-J2FwtEL+6CjGdWyZb=fBFepA@mail.gmail.com>

Which end is doing the hanging up? If it's the far end then the settings
here aren't the issue. It may be that the far end is timing out. Check this
out and give it a try:
http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
-MC

On Mon, Apr 16, 2012 at 7:02 AM, George Sapak <sat at calgaryit.com> wrote:

> I have a quite a few people that want to leave voice mails and they say
> they are being hung upon as if they were not recording a message, I think
> its usually from people that are quitter speakers, these are my settings in
> my voicemail file:
>
> <param name="record-silence-threshold" value="1"/>
> <param name="record-silence-hits" value="999"/>
>
> is there a way to set the params so the vm records event if there is
> almost total silence
>
> Thank you, George.
>
>
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From bdfoster at endigotech.com  Tue Apr 17 04:07:19 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 16 Apr 2012 20:07:19 -0400
Subject: [Freeswitch-users] Voicemail hangups
In-Reply-To: <CAKzWOxXE=QWui8=h_zaXJQsMH-J2FwtEL+6CjGdWyZb=fBFepA@mail.gmail.com>
References: <1422889126.160382.1334584443659.JavaMail.root@server3>
	<79894403.160416.1334584978542.JavaMail.root@server3>
	<CAKzWOxXE=QWui8=h_zaXJQsMH-J2FwtEL+6CjGdWyZb=fBFepA@mail.gmail.com>
Message-ID: <CAKV-jFywAp42_dqyTFay1E+UmU9GWYmsiENd3xEmWp0SXDzohg@mail.gmail.com>

I've actually had this issue too, and upgrading to the latest GIT solved
it. If that still doesn't do it for you, it's high time to file a JIRA. Bug
reports are not permitted on the users mailing list.

-BDF
"BDF" and associated trademarks ? 1989-2012 Brian Foster. All rights
reserved.
On Apr 16, 2012 6:41 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Which end is doing the hanging up? If it's the far end then the settings
> here aren't the issue. It may be that the far end is timing out. Check this
> out and give it a try:
> http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
> -MC
>
> On Mon, Apr 16, 2012 at 7:02 AM, George Sapak <sat at calgaryit.com> wrote:
>
>> I have a quite a few people that want to leave voice mails and they say
>> they are being hung upon as if they were not recording a message, I think
>> its usually from people that are quitter speakers, these are my settings in
>> my voicemail file:
>>
>> <param name="record-silence-threshold" value="1"/>
>> <param name="record-silence-hits" value="999"/>
>>
>> is there a way to set the params so the vm records event if there is
>> almost total silence
>>
>> Thank you, George.
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gabe at gundy.org  Tue Apr 17 05:08:17 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 16 Apr 2012 19:08:17 -0600
Subject: [Freeswitch-users] what is the purpose and value of the files
 in the db directory?
In-Reply-To: <CAKzWOxXFLQAFC0bYt=iCjFHoVVHs-YFOxMz9GUEGbP+9T7JGtQ@mail.gmail.com>
References: <CAE3mkNALxP5pVW9Ct3qb7KJy-k6HAhpD7vjnXGHAR1vgAUO+pA@mail.gmail.com>
	<CBAF0DC2.312AB%krice@freeswitch.org>
	<CAKzWOxXFLQAFC0bYt=iCjFHoVVHs-YFOxMz9GUEGbP+9T7JGtQ@mail.gmail.com>
Message-ID: <CAMwZ-tJMPPz-w+KRg1kbTdFenmzmvApWDq+Qn6g0AMFp82+vMg@mail.gmail.com>

On Mon, Apr 16, 2012 at 3:24 PM, Michael Collins <msc at freeswitch.org> wrote:
> FWIW, I've used the sqlite3 CLI program to query and manipulate the .db
> files. I've never run into any drama or issues.

Yeah, so have I... even before I knew the code was forked :) So, yeah,
it seems to work just fine and is a pretty helpful tool.


Best,
Gabe


From Daniel.Knaggs at realitysolutions.co.uk  Tue Apr 17 12:13:17 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Tue, 17 Apr 2012 08:13:17 +0000
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls
In-Reply-To: <CAKzWOxXQ1LCp_22r+my8ZP894kY5v3Y0gf++hosctRKy4FyCXw@mail.gmail.com>
References: <8C26A4FDAE599041A13EB499117D3C287CA27576@EX-MB-1.corp.atlasnetworks.us>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05CF91@RSV16.realitysolutions.local>
	<CAKzWOxXQ1LCp_22r+my8ZP894kY5v3Y0gf++hosctRKy4FyCXw@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05E0CA@RSV16.realitysolutions.local>

I've raised a ticket with Sangoma via this link: http://sangoma.com/support/contact_support.html

Let's see where we go...




From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: 16 April 2012 22:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

I still suspect something in FreeTDM. From Nathan's perspective that would be "inside" FreeSWITCH since that's the TDM stack we use. I highly recommend you open a trouble ticket with Sangoma and have each person who is experiencing the issue put a comment on the case and offer to supply configs, logs, and/or access that Sangoma would need to do troubleshooting.

-MC
On Fri, Apr 13, 2012 at 12:35 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk> wrote:
That's very interesting, Nathan.

Now that makes three of us, two using the Digium cards and one using a Sangoma.


It very much looks like something FreeSwitch is doing, perhaps the DTMF detection is far too sensitive? ?Are DTMF tones for an ISDN line inband or do they come in on the D-channel?


Michael - any ideas what we can all do?




From: Nathan Eisenberg [mailto:nathan at atlasnetworks.us]
Sent: 13 April 2012 00:21
To: Daniel Knaggs
Subject: [Freeswitch-users] Strange DTMF Tones On Inbound Calls

I'm having the exact issue you're describing, with the same card. ?However, what's interesting to me is that if I fire up Asterisk instead of freeswitch, the phantom DTMF problem disappears entirely.

So it's *something* inside Freeswitch, at least for me. ?Would be interesting if you're able to do a similar test.
Nathan Eisenberg
Atlas Networks | Sr. Systems Administrator
office: 206.577.3078 ?| ?www.atlasnetworks.us
?D


Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel: ? ?01482 373104
Mobile: 07932 408313
Email: ?mailto:Daniel.Knaggs at realitysolutions.co.uk
http://




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From azza.miled at gmail.com  Tue Apr 17 12:17:19 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 17 Apr 2012 09:17:19 +0100
Subject: [Freeswitch-users] FreeSWITCH and Opensips
In-Reply-To: <CAKzWOxUJqd+jHaWbjhCAVtm6XaYS3XzRtnZ1b3mf97-GqNJVMA@mail.gmail.com>
References: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
	<CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>
	<CAGQdkzRfjkbiaWuqTpC-3aHv2A-KhrLRkQ_Bw3=+Vq+9qZPf3w@mail.gmail.com>
	<CAKzWOxUJqd+jHaWbjhCAVtm6XaYS3XzRtnZ1b3mf97-GqNJVMA@mail.gmail.com>
Message-ID: <CAGQdkzT=jguP6ar92TEQv1yxMwkvFWLH8FRirRwbkg5ou=pH1g@mail.gmail.com>

Thanks a lot Michael.I am a real begginer in the SIP world and your
response was very helpful. It worked :)))
Thanks again.
2012/4/16 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Apr 16, 2012 at 8:36 AM, azza miled <azza.miled at gmail.com> wrote:
>
>> Thanks for the response, but evec with these steps, I still have the
>> error "call failed:proxy authentication required" when I dial the IVR
>> extension from an opensips' phone. Do any one know, what shall I do???
>>
>> Do you need SIP authentication between your FS box and your OpenSIPS box?
> I'm guessing not. If that's the case then I recommend opening up this file:
>
> /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml
>
> Find the tag with this opening line:
> <list name="domains" default="deny">
>
> Add a new line in there like this:
> <node type="allow" cidr="x.x.x.x/32"/>
> where x.x.x.x is the IP address of your OpenSIPS server.
>
> then go to fs_cli and issue this command: reloadacl
>
> Lastly, you'll need to route the inbound calls from OpenSIPS through your
> 'public' dialplan context because that's where ACL-allowed inbound calls
> normally get processed. You can either create your IVRs in the public
> context or you can transfer the inbound call over to the default context
> (or whatever context you are using.)
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From virbhati at gmail.com  Tue Apr 17 12:43:52 2012
From: virbhati at gmail.com (virendra bhati)
Date: Tue, 17 Apr 2012 14:13:52 +0530
Subject: [Freeswitch-users] hi
Message-ID: <CANNhuhduJ6zXvZO4GxSBs++gXx=DGM5UpynGJjoCjUkwAYR__g@mail.gmail.com>

-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
E-mail-: virbhati at gmail.com
Skype id:- virbhati2
Hyderabad(India)
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From azza.miled at gmail.com  Tue Apr 17 13:14:57 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 17 Apr 2012 10:14:57 +0100
Subject: [Freeswitch-users] transfering a call to another SIP server
Message-ID: <CAGQdkzRMHOk9q985UDsBVBPrW=W8m0-nwRReq-SbXZ6RCMMC9g@mail.gmail.com>

Hi!
I am using lua to make an IVR. I want to transfer the call to an another
SIP server(OpenSIPS). Is this possibe? cause the wiki documentation just
mention that we can transfer the call to a an another FS' extention.
Thanks for the help.
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From avi at avimarcus.net  Tue Apr 17 13:25:01 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 17 Apr 2012 12:25:01 +0300
Subject: [Freeswitch-users] transfering a call to another SIP server
In-Reply-To: <CAGQdkzRMHOk9q985UDsBVBPrW=W8m0-nwRReq-SbXZ6RCMMC9g@mail.gmail.com>
References: <CAGQdkzRMHOk9q985UDsBVBPrW=W8m0-nwRReq-SbXZ6RCMMC9g@mail.gmail.com>
Message-ID: <CALNZuwkV=YJds1t6yOHBaopBcqbQcpFB_-C6+MrMJKWfXD7N=w@mail.gmail.com>

You can bridge to an external server -- e.g. sofia/external/
1000 at myserver.com (if you left your profile named external).

Or you can take FreeSWITCH out of the call path: if you answered
deflect<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect>
sends
a REFER. If you didn't answer yet send a
redirect<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect>
which
sends a 302 temporarily moved.

-Avi


On Tue, Apr 17, 2012 at 12:14 PM, azza miled <azza.miled at gmail.com> wrote:

> Hi!
> I am using lua to make an IVR. I want to transfer the call to an another
> SIP server(OpenSIPS). Is this possibe? cause the wiki documentation just
> mention that we can transfer the call to a an another FS' extention.
> Thanks for the help.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From azza.miled at gmail.com  Tue Apr 17 13:37:00 2012
From: azza.miled at gmail.com (azza miled)
Date: Tue, 17 Apr 2012 10:37:00 +0100
Subject: [Freeswitch-users] transfering a call to another SIP server
In-Reply-To: <CALNZuwkV=YJds1t6yOHBaopBcqbQcpFB_-C6+MrMJKWfXD7N=w@mail.gmail.com>
References: <CAGQdkzRMHOk9q985UDsBVBPrW=W8m0-nwRReq-SbXZ6RCMMC9g@mail.gmail.com>
	<CALNZuwkV=YJds1t6yOHBaopBcqbQcpFB_-C6+MrMJKWfXD7N=w@mail.gmail.com>
Message-ID: <CAGQdkzT7pMt=iWNCLY-Ly-T2auE2Uz6+gCVEo6aEXR8ErH3eWQ@mail.gmail.com>

thank you, but to be clearer, I want that my lua IVR script transfer the
call to OpenSIPS. So, if I get what you said, I have to bridge to an
external server (in the xml dialplan) and then,I just have to transfer the
call to the IP address of my Opensips server in my lua script. am I wrong??

2012/4/17 Avi Marcus <avi at avimarcus.net>

> You can bridge to an external server -- e.g. sofia/external/
> 1000 at myserver.com (if you left your profile named external).
>
> Or you can take FreeSWITCH out of the call path: if you answered deflect<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect> sends
> a REFER. If you didn't answer yet send a redirect<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect> which
> sends a 302 temporarily moved.
>
> -Avi
>
>
> On Tue, Apr 17, 2012 at 12:14 PM, azza miled <azza.miled at gmail.com> wrote:
>
>> Hi!
>> I am using lua to make an IVR. I want to transfer the call to an another
>> SIP server(OpenSIPS). Is this possibe? cause the wiki documentation just
>> mention that we can transfer the call to a an another FS' extention.
>> Thanks for the help.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From miha at softnet.si  Tue Apr 17 13:41:05 2012
From: miha at softnet.si (Miha)
Date: Tue, 17 Apr 2012 11:41:05 +0200
Subject: [Freeswitch-users] Freeswitch and Opensips
Message-ID: <4F8D3AB1.1030207@softnet.si>

Hi to all,

I have configured opensips that works like load_balancer (wiki on FS page).
So, we have SBC, trunk is made to Opensips, than opensips is 
load_balacing to FSs.
When you call from FS, call is going to Opensips, then opensips redirect 
the call to SBC, if call must go outside.

My question:

FSs boxes are all the same (same dialplan, directory configuration, 
etc.). What happens if call is made to a user, which is registered on 
one of the FS box? I guess I must change dialplan and set that all 
trafic FS send to opensips (not internally), because if someone calls to 
a local user, but user is registered on other FS box, FS will behave as 
user is not registered.

So, will opensips know which user is registered on which FS box when 
opensips will be doing load_balacing?

Thanks for explenation.

Regards,
Miha
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From sherifomran2000 at yahoo.com  Tue Apr 17 13:47:36 2012
From: sherifomran2000 at yahoo.com (Sherif Omran)
Date: Tue, 17 Apr 2012 02:47:36 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <4F8D3AB1.1030207@softnet.si>
Message-ID: <1334656056.53998.YahooMailClassic@web110802.mail.gq1.yahoo.com>

Hi Miha,

I have a similar problem and would appreciate if you could give me a clue.
I ve opensips as sbc and FS but i wonder why calls do not reach fs box.

Could you please help me in configuring opensips with freeswitch as u made it?

Thank u in advance

regards,
Sherif

--- On Tue, 4/17/12, Miha <miha at softnet.si> wrote:

From: Miha <miha at softnet.si>
Subject: [Freeswitch-users] Freeswitch and Opensips
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, April 17, 2012, 11:41 AM


  

    
  
  
    Hi to all,

      

      I have configured opensips that works like load_balancer (wiki on
      FS page).
    

      So, we have SBC, trunk is made to Opensips, than opensips is
      load_balacing to FSs.

      When you call from FS, call is going to Opensips, then opensips
      redirect the call to SBC, if call must go outside.

      

      My question:
    

      

      FSs boxes are all the same (same dialplan, directory
      configuration, etc.). What happens if call is made to a user,
      which is registered on one of the FS box? I guess I must change
      dialplan and set that all trafic FS send to opensips (not
      internally), because if someone calls to a local user, but user is
      registered on other FS box, FS will behave as user is not
      registered.
    

      

      So, will opensips know which user is registered on which FS box
      when opensips will be doing load_balacing?
    

      

      Thanks for explenation.
    

      

      Regards,
    

      Miha

  


-----Inline Attachment Follows-----

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From miha at softnet.si  Tue Apr 17 14:10:00 2012
From: miha at softnet.si (Miha)
Date: Tue, 17 Apr 2012 12:10:00 +0200
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <1334656056.53998.YahooMailClassic@web110802.mail.gq1.yahoo.com>
References: <1334656056.53998.YahooMailClassic@web110802.mail.gq1.yahoo.com>
Message-ID: <4F8D4178.3040909@softnet.si>

Hi Sherif,

I hope I can give you a bit of help as my knowledge of Opensips is not 
quite good:)

First I have configured opensips as is written on FS wiki 
(http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS). Did 
you fallow steps? This shoud work and opensips should load balacing 
calls to FS box.

Than I have made like this:

if (is_method("INVITE")) {
             if($rU=~"^[0]+"){
                 if (!load_balance("1","pstn","1")) {
                         send_reply("503","Service Unavailable");
                         exit;
                 }
             }
             else{
             rewritehostport("xxx.xxx.xxx.xxx:5060");
             }


         }


Regards,
Miha

On 4/17/2012 11:47 AM, Sherif Omran wrote:
> Hi Miha,
>
> I have a similar problem and would appreciate if you could give me a clue.
> I ve opensips as sbc and FS but i wonder why calls do not reach fs box.
>
> Could you please help me in configuring opensips with freeswitch as u 
> made it?
>
> Thank u in advance
>
> regards,
> Sherif
>
> --- On *Tue, 4/17/12, Miha /<miha at softnet.si>/* wrote:
>
>
>     From: Miha <miha at softnet.si>
>     Subject: [Freeswitch-users] Freeswitch and Opensips
>     To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>     Date: Tuesday, April 17, 2012, 11:41 AM
>
>     Hi to all,
>
>     I have configured opensips that works like load_balancer (wiki on
>     FS page).
>     So, we have SBC, trunk is made to Opensips, than opensips is
>     load_balacing to FSs.
>     When you call from FS, call is going to Opensips, then opensips
>     redirect the call to SBC, if call must go outside.
>
>     My question:
>
>     FSs boxes are all the same (same dialplan, directory
>     configuration, etc.). What happens if call is made to a user,
>     which is registered on one of the FS box? I guess I must change
>     dialplan and set that all trafic FS send to opensips (not
>     internally), because if someone calls to a local user, but user is
>     registered on other FS box, FS will behave as user is not registered.
>
>     So, will opensips know which user is registered on which FS box
>     when opensips will be doing load_balacing?
>
>     Thanks for explenation.
>
>     Regards,
>     Miha
>
>     -----Inline Attachment Follows-----
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org </mc/compose?to=consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sharad at coraltele.com  Tue Apr 17 14:22:30 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 17 Apr 2012 15:52:30 +0530
Subject: [Freeswitch-users] Freeswitch and network bonding
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <4D4D700662D8429188A8259DD92622D3@sharad>

Hi All

I am trying to run the Freeswitch with network bonding. My scanerio is -

eth0    -    192.168.4.25
eth1    -    192.168.4.30
bond0    -    192.168.4.65        This is the virtual IP which is to access 
the freeswitch.

The issue is when I make a call to 192.168.4.65 which is a virtual ip of 
Freeswitch system, this call dies not reach at Freeswitch. When I do 
wireshark, wireshark shows `Destination unreachable'.

The same freeswitch works well if I disable the network bonding.

Here I would like to mention that I have not made any kind of settings in 
Freeswitch to run with network bonding.

Can someone help me why call is not arriving at Freeswitch when network 
bonding is enabled.

Thanks in advance.

Best Regards
Sharad 



From peter.olsson at visionutveckling.se  Tue Apr 17 14:31:06 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 17 Apr 2012 10:31:06 +0000
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <4D4D700662D8429188A8259DD92622D3@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>,
	<4D4D700662D8429188A8259DD92622D3@sharad>
Message-ID: <1FFF97C269757C458224B7C895F35F15092497@cantor.std.visionutv.se>

I guess it tries to listen to one of the other interfaces instead. You can force what IP to listen on, but the most obvious would maybe be to get rid of the IP's on eth0 and eth1 - since these interfaces doesn't need IP addresses anyway.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Sharad Garg [sharad at coraltele.com]
Skickat: den 17 april 2012 12:22
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Freeswitch and network bonding

Hi All

I am trying to run the Freeswitch with network bonding. My scanerio is -

eth0    -    192.168.4.25
eth1    -    192.168.4.30
bond0    -    192.168.4.65        This is the virtual IP which is to access
the freeswitch.

The issue is when I make a call to 192.168.4.65 which is a virtual ip of
Freeswitch system, this call dies not reach at Freeswitch. When I do
wireshark, wireshark shows `Destination unreachable'.

The same freeswitch works well if I disable the network bonding.

Here I would like to mention that I have not made any kind of settings in
Freeswitch to run with network bonding.

Can someone help me why call is not arriving at Freeswitch when network
bonding is enabled.

Thanks in advance.

Best Regards
Sharad


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f8d42f932761614173760!



From B.Tietz at pinguin.ag  Tue Apr 17 14:33:58 2012
From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag)
Date: Tue, 17 Apr 2012 12:33:58 +0200
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <4D4D700662D8429188A8259DD92622D3@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
	<4D4D700662D8429188A8259DD92622D3@sharad>
Message-ID: <07BF4904977CC645B485E970424193AD0FF88ECDAF@localhost>

Hi,

from http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover#Sofia_Profiles
there you have: 
'Also, you'll have to specify the SIP, RTP bind IP etc to be the floating IP, rather than the machine's auto-detected IP.
rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip'

So. change the above params in your internal profile and the bonding IP should work.

VG,
Benjamin T.

-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Sharad Garg
Gesendet: Dienstag, 17. April 2012 12:23
An: freeswitch-users at lists.freeswitch.org
Betreff: [Freeswitch-users] Freeswitch and network bonding

Hi All

I am trying to run the Freeswitch with network bonding. My scanerio is -

eth0    -    192.168.4.25
eth1    -    192.168.4.30
bond0    -    192.168.4.65        This is the virtual IP which is to access 
the freeswitch.

The issue is when I make a call to 192.168.4.65 which is a virtual ip of Freeswitch system, this call dies not reach at Freeswitch. When I do wireshark, wireshark shows `Destination unreachable'.

The same freeswitch works well if I disable the network bonding.

Here I would like to mention that I have not made any kind of settings in Freeswitch to run with network bonding.

Can someone help me why call is not arriving at Freeswitch when network bonding is enabled.

Thanks in advance.

Best Regards
Sharad 



From oseslija at gmail.com  Tue Apr 17 14:54:25 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Tue, 17 Apr 2012 12:54:25 +0200
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <4D4D700662D8429188A8259DD92622D3@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
	<4D4D700662D8429188A8259DD92622D3@sharad>
Message-ID: <CAM21H-Mkkbsjo9iW8Jg4-nHxuJWuCbLYECNc3800BWXkarWoTg@mail.gmail.com>

This scenario is wrong, there shouldn't be any IP addresses configured on
the physical interfaces.

On Tue, Apr 17, 2012 at 12:22 PM, Sharad Garg <sharad at coraltele.com> wrote:

> Hi All
>
> I am trying to run the Freeswitch with network bonding. My scanerio is -
>
> eth0    -    192.168.4.25
> eth1    -    192.168.4.30
> bond0    -    192.168.4.65        This is the virtual IP which is to access
> the freeswitch.
>
> The issue is when I make a call to 192.168.4.65 which is a virtual ip of
> Freeswitch system, this call dies not reach at Freeswitch. When I do
> wireshark, wireshark shows `Destination unreachable'.
>
> The same freeswitch works well if I disable the network bonding.
>
> Here I would like to mention that I have not made any kind of settings in
> Freeswitch to run with network bonding.
>
> Can someone help me why call is not arriving at Freeswitch when network
> bonding is enabled.
>
> Thanks in advance.
>
> Best Regards
> Sharad
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From darknesslabs at gmail.com  Tue Apr 17 15:02:26 2012
From: darknesslabs at gmail.com (Karol)
Date: Tue, 17 Apr 2012 07:02:26 -0400
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <4D4D700662D8429188A8259DD92622D3@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
	<4D4D700662D8429188A8259DD92622D3@sharad>
Message-ID: <CAFZs+v_+80Gb98LQuB4ibNvzWv_B+7czSnKrmUsqYvRmKXT9WA@mail.gmail.com>

Yeah you can't run the IP stack with doing a bond. You may want to do a
aliase on top of the bond interface. The normal interfaces act as slave to
the virtual bond interface.
On Apr 17, 2012 6:25 AM, "Sharad Garg" <sharad at coraltele.com> wrote:

> Hi All
>
> I am trying to run the Freeswitch with network bonding. My scanerio is -
>
> eth0    -    192.168.4.25
> eth1    -    192.168.4.30
> bond0    -    192.168.4.65        This is the virtual IP which is to access
> the freeswitch.
>
> The issue is when I make a call to 192.168.4.65 which is a virtual ip of
> Freeswitch system, this call dies not reach at Freeswitch. When I do
> wireshark, wireshark shows `Destination unreachable'.
>
> The same freeswitch works well if I disable the network bonding.
>
> Here I would like to mention that I have not made any kind of settings in
> Freeswitch to run with network bonding.
>
> Can someone help me why call is not arriving at Freeswitch when network
> bonding is enabled.
>
> Thanks in advance.
>
> Best Regards
> Sharad
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sharad at coraltele.com  Tue Apr 17 15:52:14 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 17 Apr 2012 17:22:14 +0530
Subject: [Freeswitch-users] Freeswitch and network bonding
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org><4D4D700662D8429188A8259DD92622D3@sharad>
	<CAM21H-Mkkbsjo9iW8Jg4-nHxuJWuCbLYECNc3800BWXkarWoTg@mail.gmail.com>
Message-ID: <980C368D72A94CC0978872726DA09E7E@sharad>

Hi All 

Thanks for your reply.

I have removed the IP from real interface. In fact both the interfaces are disbaled.

Secondly I have defined "bind_server_ip" in vars.xml.

But issue is same. 

When I do `global_getvar local_ip_v4 in CLI, it shows 127.0.0.1. Means FS is fetching the server IP as 127.0.0.1.

Plz help me to get it resolved.

Regards
Sharad
  ----- Original Message ----- 
  From: Ognjen Seslija 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, April 17, 2012 4:24 PM
  Subject: Re: [Freeswitch-users] Freeswitch and network bonding


  This scenario is wrong, there shouldn't be any IP addresses configured on the physical interfaces.


  On Tue, Apr 17, 2012 at 12:22 PM, Sharad Garg <sharad at coraltele.com> wrote:

    Hi All

    I am trying to run the Freeswitch with network bonding. My scanerio is -

    eth0    -    192.168.4.25
    eth1    -    192.168.4.30
    bond0    -    192.168.4.65        This is the virtual IP which is to access
    the freeswitch.

    The issue is when I make a call to 192.168.4.65 which is a virtual ip of
    Freeswitch system, this call dies not reach at Freeswitch. When I do
    wireshark, wireshark shows `Destination unreachable'.

    The same freeswitch works well if I disable the network bonding.

    Here I would like to mention that I have not made any kind of settings in
    Freeswitch to run with network bonding.

    Can someone help me why call is not arriving at Freeswitch when network
    bonding is enabled.

    Thanks in advance.

    Best Regards
    Sharad


    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.com

    FreeSWITCH-users mailing list
    FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Tue Apr 17 15:55:20 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 17 Apr 2012 15:55:20 +0400
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <980C368D72A94CC0978872726DA09E7E@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
	<4D4D700662D8429188A8259DD92622D3@sharad>
	<CAM21H-Mkkbsjo9iW8Jg4-nHxuJWuCbLYECNc3800BWXkarWoTg@mail.gmail.com>
	<980C368D72A94CC0978872726DA09E7E@sharad>
Message-ID: <CANBhyPCO6zKQ2UsC2RCYHaR3gdev7jmx0iDiX8Z0k5rmeGxV1g@mail.gmail.com>

Do not forget to restart internal profile.

17 ?????? 2012 ?. 15:52 ???????????? Sharad Garg <sharad at coraltele.com>???????:

> **
> Hi All
>
> Thanks for your reply.
>
> I have removed the IP from real interface. In fact both the interfaces are
> disbaled.
>
> Secondly I have defined "bind_server_ip" in vars.xml.
>
> But issue is same.
>
> When I do `global_getvar local_ip_v4 in CLI, it shows 127.0.0.1. Means FS
> is fetching the server IP as 127.0.0.1.
>
> Plz help me to get it resolved.
>
> Regards
> Sharad
>
> ----- Original Message -----
> *From:* Ognjen Seslija <oseslija at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, April 17, 2012 4:24 PM
> *Subject:* Re: [Freeswitch-users] Freeswitch and network bonding
>
> This scenario is wrong, there shouldn't be any IP addresses configured on
> the physical interfaces.
>
> On Tue, Apr 17, 2012 at 12:22 PM, Sharad Garg <sharad at coraltele.com>wrote:
>
>> Hi All
>>
>> I am trying to run the Freeswitch with network bonding. My scanerio is -
>>
>> eth0    -    192.168.4.25
>> eth1    -    192.168.4.30
>> bond0    -    192.168.4.65        This is the virtual IP which is to
>> access
>> the freeswitch.
>>
>> The issue is when I make a call to 192.168.4.65 which is a virtual ip of
>> Freeswitch system, this call dies not reach at Freeswitch. When I do
>> wireshark, wireshark shows `Destination unreachable'.
>>
>> The same freeswitch works well if I disable the network bonding.
>>
>> Here I would like to mention that I have not made any kind of settings in
>> Freeswitch to run with network bonding.
>>
>> Can someone help me why call is not arriving at Freeswitch when network
>> bonding is enabled.
>>
>> Thanks in advance.
>>
>> Best Regards
>> Sharad
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From hkalyoncu at gmail.com  Tue Apr 17 16:04:14 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 17 Apr 2012 15:04:14 +0300
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAKbxfG9q9nee7qbFVnWqZDgvvQpSHrrhqzB4Nj7XuN3QUspjQA@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
	<CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>
	<CAKbxfG9q9nee7qbFVnWqZDgvvQpSHrrhqzB4Nj7XuN3QUspjQA@mail.gmail.com>
Message-ID: <CAPf+86WeP9mnbP9Po3FoeZ44o-Du69eXS902yAVTyDSaV8DNKA@mail.gmail.com>

thank you all for your replies, esp Gabriel and Anthony for your insightful
answers.

all processes (freeswitch, mysql, apache, vs..) running on the same
machine. i think this is
the major problem. now we are adding more hw to increase the number of fs
boxes. but on the
long run, we have to try what Anthony says.


On Sat, Apr 14, 2012 at 7:02 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> If you really want the fastest dialplan possible, write a new dialplan
> module in C.
> You only have to implement 1 function, a dialplan hunt function that
> gets data from the session and builds and returns an extension object
> which is basically just a list of app/args.  You can glue in any other
> stuff you want from there.
>
> The dialplan implementations we carry in tree tend to offer a blended
> balance of performance and flexibility/extensibility to serve as both
> a demo and a practical on-ramp.
>
>
>
>
>
> On Sat, Apr 14, 2012 at 10:51 AM, Anita Hall <anita.hall at simmortel.com>
> wrote:
> > Could he use mod_odbc_query instead of xml curl to directly get his
> dialplan
> > from a DB, thus removing the webserver from the picture altogether. I
> have
> > not tried this but it looks better or am I missing something ?
> >
> > http://wiki.freeswitch.org/wiki/Mod_odbc_query
> >
> > regards,
> > Anita
> >
> >
> >
> >
> > On Fri, Apr 13, 2012 at 10:52 PM, Gabriel Gunderson <gabe at gundy.org>
> wrote:
> >>
> >> On Fri, Apr 13, 2012 at 6:37 AM, huseyin kalyoncu <hkalyoncu at gmail.com>
> >> wrote:
> >> > we are currently have two fs boxes load balanced by osips.
> >> > in each fs we have dynamic dialplan with mod_xml_curl using php &
> apache
> >> > &
> >> > mysql. this conf is doing just fine with current load (about 20 cps)
> but
> >> > with the
> >> > increasing of incoming traffic, it shows some performance issues.
> >> > we want to increase our call throughput.
> >>
> >> You don't really give us enough to go on. A few questions that I have
> >> when I read this...
> >>
> >> * Are the DB and HTTP servers running on stand alone servers, or are
> >> they running on the FS server?
> >> * What kind of hardware are they each running on? RAM, CPU, disks?
> >> * What kind of load do the servers have when you see performance issues?
> >> * What have you done to optimize your web stack (other HTTP servers
> >> etc)? Are you caching where you can?
> >> * Have you analyzed your SQL to find any slow queries?
> >>
> >> Now, I don't expect you to answer all of these in this email thread,
> >> but I'll bet if you reviewed them, you'd be able to bump your
> >> performance without having to revisit your current approach.
> >>
> >>
> >> > i searched through fs site and mailing list and came up with following
> >> > options:
> >>
> >> I'll comment on this generically, but I can't really say what each
> >> approach will (or will not) do for your current situation.
> >>
> >>
> >> > 1) using lua or (python?) to serve dialplan instead of mod_xml_curl
> >>
> >> This works. Lua is lighter so it may give you better performance than
> >> Python (this is a non-issue if you're using Python in a long-living
> >> process with event socket or mod_xml_curl). Regardless of languages,
> >> this approach has a drawback in that processing is done on the same
> >> server that's running FreeSWITCH.
> >>
> >>
> >> > 2) writing a dialplan module (something like mod_xml_curl) in c which
> >> > will do
> >> > all the db lookups etc..
> >>
> >> That seems a little extreme. Many people have scaled way up without
> >> resorting to that approach.
> >>
> >>
> >> > 3) using mod_erlang_event in outbound mode & spawning several erlang
> >> > workers to do db lookups etc..
> >>
> >> This is about the same as #4 (give or take some Erlang magic).
> >>
> >>
> >> > 4) using mod_event_socket in outbound mode & making db lookups on a
> >> > different server.
> >>
> >> This is a fine approach when controlling the call, but you still need
> >> some basic dialplan to get the calls going where they need to go. I
> >> don't see it as the right way to solve your scaling problem, but it
> >> might be part of the overall solution.
> >>
> >>
> >> It sounds like you might be giving up on mod_xml_curl too soon.
> >> Scaling HTTP is a well known problem with lots of tools available to
> >> help you. When developing for mod_xml_curl, try returning the simplest
> >> bit of dialplan you can and do all the work (logic, DB, etc.) on the
> >> HTTP server. And don't forget that in *any* scaling scenario, you will
> >> need to bring hardware, *real* hardware :)
> >>
> >> Good luck and let us know how you end up solving the problem.
> >>
> >>
> >> Best,
> >> Gabe
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From vipkilla at gmail.com  Tue Apr 17 16:11:22 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Tue, 17 Apr 2012 08:11:22 -0400
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <4F8D3AB1.1030207@softnet.si>
References: <4F8D3AB1.1030207@softnet.si>
Message-ID: <CAC-LwPMXH+qKq+_5VftdUwPh--gMPNuAG-6WXU=HkStsRHJeFg@mail.gmail.com>

> My question:
>
> FSs boxes are all the same (same dialplan, directory configuration, etc.).
> What happens if call is made to a user, which is registered on one of the FS
> box? I guess I must change dialplan and set that all trafic FS send to
> opensips (not internally), because if someone calls to a local user, but
> user is registered on other FS box, FS will behave as user is not
> registered.
>
> So, will opensips know which user is registered on which FS box when
> opensips will be doing load_balacing?
>

No, opensips will not know which user is registered to which FS box.


From anton.jugatsu at gmail.com  Tue Apr 17 16:15:11 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 17 Apr 2012 16:15:11 +0400
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <4F8D3AB1.1030207@softnet.si>
References: <4F8D3AB1.1030207@softnet.si>
Message-ID: <CANBhyPBfXzDo3f0x7VOUw+9nc0sKrGxQSMwc_LJRCC-7b1MumA@mail.gmail.com>

Good read
http://blog.2600hz.com/post/18096749028/understanding-how-fs-supports-opensips-as-a


17 ?????? 2012 ?. 13:41 ???????????? Miha <miha at softnet.si> ???????:

>  Hi to all,
>
> I have configured opensips that works like load_balancer (wiki on FS page).
> So, we have SBC, trunk is made to Opensips, than opensips is load_balacing
> to FSs.
> When you call from FS, call is going to Opensips, then opensips redirect
> the call to SBC, if call must go outside.
>
> My question:
>
> FSs boxes are all the same (same dialplan, directory configuration, etc.).
> What happens if call is made to a user, which is registered on one of the
> FS box? I guess I must change dialplan and set that all trafic FS send to
> opensips (not internally), because if someone calls to a local user, but
> user is registered on other FS box, FS will behave as user is not
> registered.
>
> So, will opensips know which user is registered on which FS box when
> opensips will be doing load_balacing?
>
> Thanks for explenation.
>
> Regards,
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fieldpeak at gmail.com  Tue Apr 17 17:05:05 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 17 Apr 2012 21:05:05 +0800
Subject: [Freeswitch-users] Pass the Regiseration message from SIP UA to
 remote SIP server
In-Reply-To: <CAKDfjgfFJy_O=-X1semmXcoRHj=L36k79vTns9-uEFQfJJEjLg@mail.gmail.com>
References: <CAOTmbMx5uLmL2hQfCJ785zY=hBijGW-p8WB_uM96sF=tpUpboQ@mail.gmail.com>
	<CAKV-jFyVsBRB3uKTTSME7FKiGnJrEvTj5AKAyy5uo+FLxcXACQ@mail.gmail.com>
	<CAJUJwtgvK6mrnw-mGs9fMipOxNN1sQrwiatK=kMU0AtkrEV94Q@mail.gmail.com>
	<CAKDfjgfFJy_O=-X1semmXcoRHj=L36k79vTns9-uEFQfJJEjLg@mail.gmail.com>
Message-ID: <CAOTmbMwa5497OTh0TKXbA-SGinzWDMVGZvbQ7CHaQScPyUaDLQ@mail.gmail.com>

Yes. got it, thanks all for kindly reply.
Cheers!

BR,
Charles

2012/4/12 Kristian Kielhofner <kris at kriskinc.com>

> Precisely.  The OP should put a SER-something in front of his
> FreeSWITCH instances and use the path module.  FreeSWITCH supports
> receiving REGISTER requests with path support but it doesn't support
> sending them (which makes perfect sense given its intended function).
>
> On Wed, Apr 11, 2012 at 3:57 PM, SamyGo <govoiper at gmail.com> wrote:
> > Hello,
> > What i've understood is that you want the REGISTER requests to be
> relayed to
> > some other server via FreeSWITCH ! I'd say use OpenSIPS or Kamailio (SIP
> > proxies) to do this kind of magic.
> >
> > Regards,
> > Sammy.
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Charles
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From hkalyoncu at gmail.com  Tue Apr 17 17:15:24 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 17 Apr 2012 16:15:24 +0300
Subject: [Freeswitch-users] mod_com_g729 log file
Message-ID: <CAPf+86Xj_-xBCNM3eiYFRW08ULjH2-97pa1g4XXmxfaA4-EmDg@mail.gmail.com>

hello,

mod_com_g729 creates a log file named "1" under /usr/local/freeswitch which
is containing lost of lines just like these:

Success allocating G.729A/1
Success allocating G.729A/0
Success deallocating G.729A/1
Success deallocating G.729A/0
Success deallocating G.729A/1


is there any way to disable logging for mod_com_g729?


regards,
huseyin
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From sat at calgaryit.com  Tue Apr 17 17:17:04 2012
From: sat at calgaryit.com (George Sapak)
Date: Tue, 17 Apr 2012 07:17:04 -0600 (MDT)
Subject: [Freeswitch-users] Voicemail hangups
In-Reply-To: <CAKV-jFywAp42_dqyTFay1E+UmU9GWYmsiENd3xEmWp0SXDzohg@mail.gmail.com>
Message-ID: <303902946.164057.1334668624705.JavaMail.root@server3>

this is my version:

FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600)

Thank You,
George

----- Original Message -----
From: "Brian Foster" <bdfoster at endigotech.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Monday, April 16, 2012 6:07:19 PM
Subject: Re: [Freeswitch-users] Voicemail hangups




I've actually had this issue too, and upgrading to the latest GIT solved it. If that still doesn't do it for you, it's high time to file a JIRA. Bug reports are not permitted on the users mailing list. 

-BDF 
"BDF" and associated trademarks ? 1989-2012 Brian Foster. All rights reserved. 
On Apr 16, 2012 6:41 PM, "Michael Collins" < msc at freeswitch.org > wrote: 


Which end is doing the hanging up? If it's the far end then the settings here aren't the issue. It may be that the far end is timing out. Check this out and give it a try: 
http://wiki.freeswitch.org/wiki/Variable_record_waste_resources 
-MC 


On Mon, Apr 16, 2012 at 7:02 AM, George Sapak < sat at calgaryit.com > wrote: 


I have a quite a few people that want to leave voice mails and they say they are being hung upon as if they were not recording a message, I think its usually from people that are quitter speakers, these are my settings in my voicemail file: 

<param name="record-silence-threshold" value="1"/> 
<param name="record-silence-hits" value="999"/> 

is there a way to set the params so the vm records event if there is almost total silence 

Thank you, George. 


_________________________________________________________________________ 
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org 
http://www.freeswitchsolutions.com 

 
 

Official FreeSWITCH Sites 
http://www.freeswitch.org 
http://wiki.freeswitch.org 
http://www.cluecon.com 

FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From fieldpeak at gmail.com  Tue Apr 17 17:23:47 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 17 Apr 2012 21:23:47 +0800
Subject: [Freeswitch-users] Call transfer failed when bypass media
Message-ID: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>

Dear AM,

i would like to make media bypass for lower FS traffic, however, if i
enable media bypass, the call transfer failed, the parameters as following,

   <param name="inbound-bypass-media" value="true"/>
    <param name="inbound-proxy-media" value="false"/>

but if i disable media bypass like configuration below, call transfer works
well.

   <param name="inbound-bypass-media" value="false"/>
    <param name="inbound-proxy-media" value="false"/>

the scenario is very simple, PSTN user X->media GW->FS->Registered user A,
A answer the call, then press hold button and dial B, A talked with B, A
hangup the call, pstn user X should talk with B.

Could you please advise if anything else i missed, i below FS so strong, it
must support this scenario, i was suffered for this issue for over two
years... thanks in advance.

-- 
Regards,
Charles
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From fieldpeak at gmail.com  Tue Apr 17 17:27:55 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 17 Apr 2012 21:27:55 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for iPhone
Message-ID: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>

Is there any recommended free/open source sip client for iPhone? Thanks.

-- 
Regards,
Charles
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From anton.vazir at gmail.com  Tue Apr 17 17:40:21 2012
From: anton.vazir at gmail.com (Anton VG)
Date: Tue, 17 Apr 2012 18:40:21 +0500
Subject: [Freeswitch-users] creating a conference for 2 separate parked
 UUIDs, please help
In-Reply-To: <CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>
References: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
	<CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>
Message-ID: <CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>

Thanks Michael!

Actually I've already managed to do it via dialplan, and I was mostly
thinking how to do it without dialplan at all,
seems in conferencing case dialplan cannot be avoided,

Regards,
Anton

2012/4/17 Michael Collins <msc at freeswitch.org>:
> Conferences get created on the fly. For example, in the default configs you
> can call 3000 and it magically creates a conference named 3000-${domain} on
> your system. For testing, try doing a uuid_transfer of these parked calls
> and drop them into the dialplan that in turn sends them to the appropriate
> conference:
>
> uuid_transfer <uuid> 3000
> or
> uuid_transfer <uuid> 3000 XML default
>
> Roll your own version of x3000 if you want the conference to be
> unique/special/different.
>
> -MC
>
> On Sun, Apr 15, 2012 at 4:11 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Guys,
>>
>> I need to create a conference for 2 UUID's , which already exists and
>> parked. How do I create a new, empty conference and then send those 2
>> UUIDs there?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fs-list at communicatefreely.net  Tue Apr 17 17:49:01 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 17 Apr 2012 09:49:01 -0400
Subject: [Freeswitch-users] SIP trace specific call or uuid
In-Reply-To: <CALNZuwmW7Y1yNm7uQDUzV1ZVGxm5EjUsByQ86azx1vrwcSjReQ@mail.gmail.com>
References: <CAGX+XzsboC5mUU_SzVtc3BMDRpGJiC7haaGRgNL-6R=mRp0jYw@mail.gmail.com>
	<CALNZuwmW7Y1yNm7uQDUzV1ZVGxm5EjUsByQ86azx1vrwcSjReQ@mail.gmail.com>
Message-ID: <4F8D74CD.3040606@communicatefreely.net>

Yes, I wish there was a way too.

In the interim, I use sofia status profile user XXX to find out their IP 
and port, then on the same machine, I use tcpdump to capture the data.

tcpdump -i INTERFACE -s 1500 -A host IPADDRESS and port SIPPORT will get 
you what you want.

You can also use tcpdump -i INTERFACE -s 1500 -w my_capture.pcap host 
IPADDRESS and port SIPPORT to generate a file that can be opened in 
wireshark.  This is much easier to work with.

Even if you capture everything on the interface, wireshark can sort out 
the individual calls, so it can still be useful.

-Tim

Avi Marcus wrote:
> Not within FS, yet.
> Try ngrep - http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep
>  
> -Avi
>
>
>
> On Fri, Mar 30, 2012 at 11:59 AM, Anton VG <anton.vazir at gmail.com 
> <mailto:anton.vazir at gmail.com>> wrote:
>
>     HI!
>
>     Is there a way to enable sip-trace on the specific call or uuid?
>
>     I've been only able to find how to enable it per profile, but on a
>     busy situation, the information amount
>     is too high to be useful, so it's often necessary to trace a specific
>     call or call to specific destination.
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From peter.olsson at visionutveckling.se  Tue Apr 17 17:50:29 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 17 Apr 2012 13:50:29 +0000
Subject: [Freeswitch-users] creating a conference for 2 separate parked
 UUIDs, please help
In-Reply-To: <CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>
References: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
	<CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>,
	<CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>
Message-ID: <1FFF97C269757C458224B7C895F35F1509269C@cantor.std.visionutv.se>

If you're using ESL you can just use sendmsg/execute and execute the conference application directly on the call legs.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anton VG [anton.vazir at gmail.com]
Skickat: den 17 april 2012 15:40
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] creating a conference for 2 separate parked UUIDs, please help

Thanks Michael!

Actually I've already managed to do it via dialplan, and I was mostly
thinking how to do it without dialplan at all,
seems in conferencing case dialplan cannot be avoided,

Regards,
Anton

2012/4/17 Michael Collins <msc at freeswitch.org>:
> Conferences get created on the fly. For example, in the default configs you
> can call 3000 and it magically creates a conference named 3000-${domain} on
> your system. For testing, try doing a uuid_transfer of these parked calls
> and drop them into the dialplan that in turn sends them to the appropriate
> conference:
>
> uuid_transfer <uuid> 3000
> or
> uuid_transfer <uuid> 3000 XML default
>
> Roll your own version of x3000 if you want the conference to be
> unique/special/different.
>
> -MC
>
> On Sun, Apr 15, 2012 at 4:11 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Guys,
>>
>> I need to create a conference for 2 UUID's , which already exists and
>> parked. How do I create a new, empty conference and then send those 2
>> UUIDs there?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f8d719432761217048912!



From freeswitch-list at puzzled.xs4all.nl  Tue Apr 17 18:01:17 2012
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 17 Apr 2012 16:01:17 +0200
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <CAPf+86WeP9mnbP9Po3FoeZ44o-Du69eXS902yAVTyDSaV8DNKA@mail.gmail.com>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
	<CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>
	<CAKbxfG9q9nee7qbFVnWqZDgvvQpSHrrhqzB4Nj7XuN3QUspjQA@mail.gmail.com>
	<CAPf+86WeP9mnbP9Po3FoeZ44o-Du69eXS902yAVTyDSaV8DNKA@mail.gmail.com>
Message-ID: <4F8D77AD.8010800@puzzled.xs4all.nl>

On 04/17/2012 02:04 PM, huseyin kalyoncu wrote:
> thank you all for your replies, esp Gabriel and Anthony for your
> insightful answers.
>
> all processes (freeswitch, mysql, apache, vs..) running on the same
> machine.

If it fits your requirements then you might want to look at nginx or 
lighttpd which use less resources than apache.

Regards,
Patrick


From ifoundthetao at gmail.com  Tue Apr 17 18:06:16 2012
From: ifoundthetao at gmail.com (Timothy Bolton)
Date: Tue, 17 Apr 2012 09:06:16 -0500
Subject: [Freeswitch-users] dynamic dialplan alternatives
In-Reply-To: <4F8D77AD.8010800@puzzled.xs4all.nl>
References: <CAPf+86X2yQhTRBg-zQM32CDemyzLWnYAWZGWXnAm-kWVJmK5SQ@mail.gmail.com>
	<CAMwZ-t+6-_CW8JnX_hNvOt6QYPf-p2Lv6OWV8WXKMNkb8O+66g@mail.gmail.com>
	<CAGKPtBsnQVmFMwTGgep2Nk0ciMMTgX4zLkUx7Jkrv0_5_6ZdOw@mail.gmail.com>
	<CAKbxfG9q9nee7qbFVnWqZDgvvQpSHrrhqzB4Nj7XuN3QUspjQA@mail.gmail.com>
	<CAPf+86WeP9mnbP9Po3FoeZ44o-Du69eXS902yAVTyDSaV8DNKA@mail.gmail.com>
	<4F8D77AD.8010800@puzzled.xs4all.nl>
Message-ID: <4F8D78D8.3060102@gmail.com>

Also, look into GWAN.  It's incredibly fast, has a tiny memory 
footprint, and you write your scripts in ANSI C.  I think OS may only be 
an issue if you are using Windows or BSD.


'We who cut mere stones must always be envisioning cathedrals.'
Quarry Worker's Creed


On 4/17/2012 9:01 AM, Patrick Lists wrote:
> On 04/17/2012 02:04 PM, huseyin kalyoncu wrote:
>> thank you all for your replies, esp Gabriel and Anthony for your
>> insightful answers.
>>
>> all processes (freeswitch, mysql, apache, vs..) running on the same
>> machine.
> If it fits your requirements then you might want to look at nginx or
> lighttpd which use less resources than apache.
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anton.jugatsu at gmail.com  Tue Apr 17 18:25:08 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 17 Apr 2012 18:25:08 +0400
Subject: [Freeswitch-users] SIP trace specific call or uuid
In-Reply-To: <4F8D74CD.3040606@communicatefreely.net>
References: <CAGX+XzsboC5mUU_SzVtc3BMDRpGJiC7haaGRgNL-6R=mRp0jYw@mail.gmail.com>
	<CALNZuwmW7Y1yNm7uQDUzV1ZVGxm5EjUsByQ86azx1vrwcSjReQ@mail.gmail.com>
	<4F8D74CD.3040606@communicatefreely.net>
Message-ID: <CANBhyPDBNDbrQJTb8SF7KkPuYVzNkVGiCR2QVPi1GqcLbc7mbA@mail.gmail.com>

Also you can use pcapsipdump http://pcapsipdump.sourceforge.net/ or even
 homer http://www.sipcapture.org/

17 ?????? 2012 ?. 17:49 ???????????? Tim St. Pierre <
fs-list at communicatefreely.net> ???????:

> Yes, I wish there was a way too.
>
> In the interim, I use sofia status profile user XXX to find out their IP
> and port, then on the same machine, I use tcpdump to capture the data.
>
> tcpdump -i INTERFACE -s 1500 -A host IPADDRESS and port SIPPORT will get
> you what you want.
>
> You can also use tcpdump -i INTERFACE -s 1500 -w my_capture.pcap host
> IPADDRESS and port SIPPORT to generate a file that can be opened in
> wireshark.  This is much easier to work with.
>
> Even if you capture everything on the interface, wireshark can sort out
> the individual calls, so it can still be useful.
>
> -Tim
>
> Avi Marcus wrote:
> > Not within FS, yet.
> > Try ngrep - http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep
> >
> > -Avi
> >
> >
> >
> > On Fri, Mar 30, 2012 at 11:59 AM, Anton VG <anton.vazir at gmail.com
> > <mailto:anton.vazir at gmail.com>> wrote:
> >
> >     HI!
> >
> >     Is there a way to enable sip-trace on the specific call or uuid?
> >
> >     I've been only able to find how to enable it per profile, but on a
> >     busy situation, the information amount
> >     is too high to be useful, so it's often necessary to trace a specific
> >     call or call to specific destination.
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From david at harktech.com  Tue Apr 17 19:15:39 2012
From: david at harktech.com (David George)
Date: Tue, 17 Apr 2012 11:15:39 -0400
Subject: [Freeswitch-users] freetdm config question
Message-ID: <4F8D891B.4070200@harktech.com>

Hello all,

I just had a quick question on FreeTDM configuration.  I have looked 
through the Wiki (both FreeSWITCH and Sangoma) and found a couple of 
references to running T1 boards with robbed bit signaling (RBS), but I 
have not gotten any configuration to work.

Is anyone using FreeTDM with T1 RBS and able to comment on my freetdm.conf?

[span zt tsu600e]
fxo-channel =>  1-4

I am using an Adtran channel bank (TSU600e) with FXS cards and a 
telephone to try to generate test calls (T1 configured as AMI/D4).  This 
is connected to a Digium TE-220 Dual T1 card.  I am using the Dahdi 
2.6.0 drivers and I have no alarms on the T1.

I am also using the latest FreeSwitch git (as of Apr 17 10am Eastern 
Daylight Time).

Any help or pointers would be appreciated.

Thanks,

-- 
David George
Hark Technologies



From yehavi.bourvine at gmail.com  Tue Apr 17 19:22:08 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 17 Apr 2012 18:22:08 +0300
Subject: [Freeswitch-users] how to see which extensions are defined?
Message-ID: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>

Hello,

  We have the following odd thing: I've tried updating FreeSwitch version
on my test machine two days ago and since then no extension can be
registered: they all get "403 fobidden". Going back to an older version
makes it work.

I am missing some tool/command to see which extensions are defined (i.e.
something similar to "sofia status profile XXX reg", but for defined, not
only registered extensions). I would like to make sure that it reads
correctly the extensions definitions.

Any such thing?

I do not open JIRA yet as I am not sure whether the problem is with our
config or not.

                       Thanks, __Yehavi:
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From avi at avimarcus.net  Tue Apr 17 19:57:32 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 17 Apr 2012 18:57:32 +0300
Subject: [Freeswitch-users] how to see which extensions are defined?
In-Reply-To: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
References: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
Message-ID: <CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>

If you include all your config files, then FS will complain about xml parse
errors if there are any.
So how about you try a simple grep of the config "directory" directory for
the extension -- does that help at all?

-Avi


On Tue, Apr 17, 2012 at 6:22 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello,
>
>   We have the following odd thing: I've tried updating FreeSwitch version
> on my test machine two days ago and since then no extension can be
> registered: they all get "403 fobidden". Going back to an older version
> makes it work.
>
> I am missing some tool/command to see which extensions are defined (i.e.
> something similar to "sofia status profile XXX reg", but for defined, not
> only registered extensions). I would like to make sure that it reads
> correctly the extensions definitions.
>
> Any such thing?
>
> I do not open JIRA yet as I am not sure whether the problem is with our
> config or not.
>
>                        Thanks, __Yehavi:
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 17 20:19:23 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 Apr 2012 09:19:23 -0700
Subject: [Freeswitch-users] FreeSWITCH and Opensips
In-Reply-To: <CAGQdkzT=jguP6ar92TEQv1yxMwkvFWLH8FRirRwbkg5ou=pH1g@mail.gmail.com>
References: <CAGQdkzQyJj9hLE4ny9AnpJr0HtuypEcJb61D8Ncs9E_p8s3W+Q@mail.gmail.com>
	<CAFArqsZsTH=n-FvS+54B9s9E+U5BfGGsgfdW_sANGkuO_n4ycw@mail.gmail.com>
	<CAGQdkzRfjkbiaWuqTpC-3aHv2A-KhrLRkQ_Bw3=+Vq+9qZPf3w@mail.gmail.com>
	<CAKzWOxUJqd+jHaWbjhCAVtm6XaYS3XzRtnZ1b3mf97-GqNJVMA@mail.gmail.com>
	<CAGQdkzT=jguP6ar92TEQv1yxMwkvFWLH8FRirRwbkg5ou=pH1g@mail.gmail.com>
Message-ID: <CAKzWOxVmnnF7Ojm=V9hcFpydhC+WweyWKrwJD=g2hLhotpkJmA@mail.gmail.com>

On Tue, Apr 17, 2012 at 1:17 AM, azza miled <azza.miled at gmail.com> wrote:

> Thanks a lot Michael.I am a real begginer in the SIP world and your
> response was very helpful. It worked :)))
> Thanks again.
>
Thanks for checking in. Don't forget to buy the books that I co-authored
with several other really smart guys. :)
The bridge<http://www.amazon.com/FreeSWITCH-1-0-6-Anthony-Minessale/dp/1847199968/ref=sr_1_1?ie=UTF8&qid=1334678891&sr=8-1>book
and the
cookbook<http://www.amazon.com/FreeSWITCH-Cookbook-Anthony-Minessale/dp/1849515409/ref=sr_1_2?ie=UTF8&qid=1334678891&sr=8-2>
.

-MC
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From rajesh.mbm at gmail.com  Tue Apr 17 17:23:44 2012
From: rajesh.mbm at gmail.com (Rajesh Choudhary)
Date: Tue, 17 Apr 2012 18:53:44 +0530
Subject: [Freeswitch-users] FreeSwitch Configuration files
Message-ID: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>

Hello

I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04 LTS
as per installation guide, but there are no conf files in
/usr/local/freeswitch/conf folder.However when  i run sofia profile
external restart reloadxml in FS cli itruns successfully from where it is
getting conf files kindly help

regards
Rajesh
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From barnyritchley at hotmail.com  Tue Apr 17 17:46:53 2012
From: barnyritchley at hotmail.com (Barnaby Ritchley)
Date: Tue, 17 Apr 2012 14:46:53 +0100
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>

Hi Guys

Just found that when using XML CDR module, if the script that is called
hangs, this will also hang the b-leg.  We had a situation where we were
sending CDR data to humbuglabs and because their server was responding
slowly, and in some cases not at all, this was causing b legs in freeswitch
to hang.

we added a timeout to the curl post to humbug which has solved the issue.

More of a note to be added in to the wiki for others for future.

Cheers



From barnyritchley at hotmail.com  Tue Apr 17 17:56:37 2012
From: barnyritchley at hotmail.com (Barnaby Ritchley)
Date: Tue, 17 Apr 2012 14:56:37 +0100
Subject: [Freeswitch-users] SIP trace specific call or uuid
In-Reply-To: <4F8D74CD.3040606@communicatefreely.net>
References: <CAGX+XzsboC5mUU_SzVtc3BMDRpGJiC7haaGRgNL-6R=mRp0jYw@mail.gmail.com>
	<CALNZuwmW7Y1yNm7uQDUzV1ZVGxm5EjUsByQ86azx1vrwcSjReQ@mail.gmail.com>
	<4F8D74CD.3040606@communicatefreely.net>
Message-ID: <BLU0-SMTP360E3CB3025065987FF9CA3BD3F0@phx.gbl>

Another option if you just want to view sip messaging is to use ngrep.

ngrep is useful for debugging certain calls as you can filter, ie if you wanted to view calls to and from 0123456789 you can do the following:

ngrep -qtW byline 12345678 port 5060

you can replace the phone number with ip addresses also if you wish



On 17 Apr 2012, at 14:49, Tim St. Pierre wrote:

Yes, I wish there was a way too.

In the interim, I use sofia status profile user XXX to find out their IP 
and port, then on the same machine, I use tcpdump to capture the data.

tcpdump -i INTERFACE -s 1500 -A host IPADDRESS and port SIPPORT will get 
you what you want.

You can also use tcpdump -i INTERFACE -s 1500 -w my_capture.pcap host 
IPADDRESS and port SIPPORT to generate a file that can be opened in 
wireshark.  This is much easier to work with.

Even if you capture everything on the interface, wireshark can sort out 
the individual calls, so it can still be useful.

-Tim

Avi Marcus wrote:
> Not within FS, yet.
> Try ngrep - http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep
> 
> -Avi
> 
> 
> 
> On Fri, Mar 30, 2012 at 11:59 AM, Anton VG <anton.vazir at gmail.com 
> <mailto:anton.vazir at gmail.com>> wrote:
> 
>    HI!
> 
>    Is there a way to enable sip-trace on the specific call or uuid?
> 
>    I've been only able to find how to enable it per profile, but on a
>    busy situation, the information amount
>    is too high to be useful, so it's often necessary to trace a specific
>    call or call to specific destination.
> 
>    _________________________________________________________________________
>    Professional FreeSWITCH Consulting Services:
>    consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>    http://www.freeswitchsolutions.com
> 
>    
>    
> 
>    Official FreeSWITCH Sites
>    http://www.freeswitch.org
>    http://wiki.freeswitch.org
>    http://www.cluecon.com
> 
>    FreeSWITCH-users mailing list
>    FreeSWITCH-users at lists.freeswitch.org
>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>    http://www.freeswitch.org
> 
> 
> ------------------------------------------------------------------------
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




From krice at freeswitch.org  Tue Apr 17 21:12:15 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 17 Apr 2012 12:12:15 -0500
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
Message-ID: <CBB30E9F.313CB%krice@freeswitch.org>

They should be in /usr/local/freeswitch/conf unless you used something that
would put them in /etc/freeswitch


On 4/17/12 8:23 AM, "Rajesh Choudhary" <rajesh.mbm at gmail.com> wrote:

> Hello
> 
> I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04 LTS as
> per installation guide, but there are no conf files in
> /usr/local/freeswitch/conf folder.However when? i run sofia profile external
> restart reloadxml in FS cli itruns successfully from where it is getting conf
> files kindly help
> 
> regards
> Rajesh
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sharad at coraltele.com  Tue Apr 17 20:17:45 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Tue, 17 Apr 2012 21:47:45 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 70, Issue 121
References: <mailman.7413.1334664306.13443.freeswitch-users@lists.freeswitch.org>
Message-ID: <A31C5F4EB28B4265ABCFD5B8E81528E8@sharad>

Thanks for the suggestion...

After defining the sip-ip & rtp-ip, call is arriving at Freeswitch & call is 
answered. But unfortnately, there is one way voice. FS to network side voice 
is OK but network to FS there is no voice.

any suggestion for this issue also.

Best Regards
Sharad

----- Original Message ----- 
From: <freeswitch-users-request at lists.freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, April 17, 2012 5:35 PM
Subject: FreeSWITCH-users Digest, Vol 70, Issue 121


> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>


--------------------------------------------------------------------------------


> Today's Topics:
>
>   1. Re: Freeswitch and network bonding (Karol)
>   2. Re: Freeswitch and network bonding (Sharad Garg)
>   3. Re: Freeswitch and network bonding (Anton Kvashenkin)
>   4. Re: dynamic dialplan alternatives (huseyin kalyoncu)
>


--------------------------------------------------------------------------------


> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From yehavi.bourvine at gmail.com  Tue Apr 17 22:16:01 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 17 Apr 2012 21:16:01 +0300
Subject: [Freeswitch-users] how to see which extensions are defined?
In-Reply-To: <CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>
References: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
	<CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>
Message-ID: <CAAtWKeHhCfXsPh6uaZEs57Mv9RSj42ZPcj=ni-uq3_F43bOfRw@mail.gmail.com>

FS does not complain, and the extension file is in the directory file...

                  Thanks, __Yehavi:


2012/4/17 Avi Marcus <avi at avimarcus.net>

> If you include all your config files, then FS will complain about xml
> parse errors if there are any.
> So how about you try a simple grep of the config "directory" directory for
> the extension -- does that help at all?
>
> -Avi
>
>
> On Tue, Apr 17, 2012 at 6:22 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Hello,
>>
>>   We have the following odd thing: I've tried updating FreeSwitch version
>> on my test machine two days ago and since then no extension can be
>> registered: they all get "403 fobidden". Going back to an older version
>> makes it work.
>>
>> I am missing some tool/command to see which extensions are defined (i.e.
>> something similar to "sofia status profile XXX reg", but for defined, not
>> only registered extensions). I would like to make sure that it reads
>> correctly the extensions definitions.
>>
>> Any such thing?
>>
>> I do not open JIRA yet as I am not sure whether the problem is with our
>> config or not.
>>
>>                        Thanks, __Yehavi:
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Tue Apr 17 22:45:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 Apr 2012 11:45:19 -0700
Subject: [Freeswitch-users] transfering a call to another SIP server
In-Reply-To: <CAGQdkzT7pMt=iWNCLY-Ly-T2auE2Uz6+gCVEo6aEXR8ErH3eWQ@mail.gmail.com>
References: <CAGQdkzRMHOk9q985UDsBVBPrW=W8m0-nwRReq-SbXZ6RCMMC9g@mail.gmail.com>
	<CALNZuwkV=YJds1t6yOHBaopBcqbQcpFB_-C6+MrMJKWfXD7N=w@mail.gmail.com>
	<CAGQdkzT7pMt=iWNCLY-Ly-T2auE2Uz6+gCVEo6aEXR8ErH3eWQ@mail.gmail.com>
Message-ID: <CAKzWOxVqLsnEeyHS6v4NnVRHweHjw9bYKzK9dMGCnqMZa8urPQ@mail.gmail.com>

On Tue, Apr 17, 2012 at 2:37 AM, azza miled <azza.miled at gmail.com> wrote:

> thank you, but to be clearer, I want that my lua IVR script transfer the
> call to OpenSIPS. So, if I get what you said, I have to bridge to an
> external server (in the xml dialplan) and then,I just have to transfer the
> call to the IP address of my Opensips server in my lua script. am I wrong??
>

Just be careful when using the word "transfer." It can mean two related but
very different things. There a transfer like transferring a phone call from
one phone to another. There's also a dialplan transfer application that
sends the call flow to another part of the dialplan. I suspect you'll be
doing a little of both. :)

In your Lua script I suggest that the end of the program be transferring
(i.e. dialplan transfer) to an extension in the dialplan that would then
execute the bridge over to your OpenSIPS server. Read this part of the wiki
carefully, especially the note about autohangup:

http://wiki.freeswitch.org/wiki/Lua#session:transfer

The line could be something like this:

session:transfer(target_number,"XML","to_opensips")

The dialplan extension to which you transfer the call can then just do the
bridge. You could even create an entirely new dialplan context. E.g., you
could create a file /usr/local/freeswitch/conf/dialplan/to_opensips.xml
with something like this:

<include>
  <context name="to_opensips">
    <extension name="bridge to opensips">
      <condition field="destination_number" expression="(.*)">
        <action application="bridge" data="sofia/internal/$1@
${opensips_ip_address}"/>
      </condition>
    </extension>
  </context>
</include>

I put my usual disclaimer here: This is off the top of my head. I did not
try this code. If you try it and it doesn't work then tinker and see if you
can figure it out. :)

-MC
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From msc at freeswitch.org  Tue Apr 17 22:51:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 Apr 2012 11:51:44 -0700
Subject: [Freeswitch-users] creating a conference for 2 separate parked
 UUIDs, please help
In-Reply-To: <CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>
References: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
	<CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>
	<CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>
Message-ID: <CAKzWOxVNLhNkKJCJwy=FhoTY55BCpT13bK9QLkDE1-qHGWkkBA@mail.gmail.com>

On Tue, Apr 17, 2012 at 6:40 AM, Anton VG <anton.vazir at gmail.com> wrote:

> Thanks Michael!
>
> Actually I've already managed to do it via dialplan, and I was mostly
> thinking how to do it without dialplan at all,
> seems in conferencing case dialplan cannot be avoided,
>

Challenge accepted!

uuid_transfer <uuid> conference:3000 at default inline

Usual disclaimer applies - please tinker and test.
-MC
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From msc at freeswitch.org  Wed Apr 18 00:10:35 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 17 Apr 2012 13:10:35 -0700
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
Message-ID: <CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>

Thanks for chiming in on this one. If you haven't already updated the wiki
we'll conscript someone to do it tomorrow on our conference call.
-MC

On Tue, Apr 17, 2012 at 6:46 AM, Barnaby Ritchley <barnyritchley at hotmail.com
> wrote:

> Hi Guys
>
> Just found that when using XML CDR module, if the script that is called
> hangs, this will also hang the b-leg.  We had a situation where we were
> sending CDR data to humbuglabs and because their server was responding
> slowly, and in some cases not at all, this was causing b legs in freeswitch
> to hang.
>
> we added a timeout to the curl post to humbug which has solved the issue.
>
> More of a note to be added in to the wiki for others for future.
>
> Cheers
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gabe at gundy.org  Wed Apr 18 00:16:45 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 17 Apr 2012 14:16:45 -0600
Subject: [Freeswitch-users] hi
In-Reply-To: <CANNhuhduJ6zXvZO4GxSBs++gXx=DGM5UpynGJjoCjUkwAYR__g@mail.gmail.com>
References: <CANNhuhduJ6zXvZO4GxSBs++gXx=DGM5UpynGJjoCjUkwAYR__g@mail.gmail.com>
Message-ID: <CAMwZ-t+9gx+iJ-DhckT2oF0gXo7V7ZgOh-2XZWucMo+SrtVkEQ@mail.gmail.com>

On Tue, Apr 17, 2012 at 2:43 AM, virendra bhati <virbhati at gmail.com> wrote:
> Thanks and regards

You don't give us much to go on here... Tell us, why do you use
FreeSWITCH? What do you hope to get from the mailing list? What do you
want to give back to the community :)

Oh, 'Hi" back at you.


Best,
Gabe


From gabe at gundy.org  Wed Apr 18 00:19:13 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 17 Apr 2012 14:19:13 -0600
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
Message-ID: <CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>

On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com> wrote:
> I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04 LTS
> as per installation guide, but there are no conf files in
> /usr/local/freeswitch/conf folder.

They might be in /opt/freeswitch/conf/


> However when? i run sofia profile external
> restart reloadxml in FS cli itruns successfully from where it is getting
> conf files kindly help

While FreeSWITCH is up, run this command: 'sudo lsof | grep
freeswitch' and see what files it has open.


Best,
Gabe


From gabe at gundy.org  Wed Apr 18 00:35:05 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 17 Apr 2012 14:35:05 -0600
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 70, Issue 121
In-Reply-To: <A31C5F4EB28B4265ABCFD5B8E81528E8@sharad>
References: <mailman.7413.1334664306.13443.freeswitch-users@lists.freeswitch.org>
	<A31C5F4EB28B4265ABCFD5B8E81528E8@sharad>
Message-ID: <CAMwZ-t+W4SO6THSoJSn0WX1-nUxdL8_492UTpbyJOTzJuUmJ=w@mail.gmail.com>

On Tue, Apr 17, 2012 at 10:17 AM, Sharad Garg <sharad at coraltele.com> wrote:
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."

^^^ I'd follow that advice... we have no idea what issue you're
replying too. There is no context.

Good luck.


Gabe


From mthakershi at gmail.com  Wed Apr 18 03:30:45 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 17 Apr 2012 18:30:45 -0500
Subject: [Freeswitch-users] ESL code question
Message-ID: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>

Hello,

Here is a code snippet from ESL example in wiki.
http://wiki.freeswitch.org/wiki/Esl#Examples

my $uuid = $con->api("create_uuid")->getBody();
my $res =
    $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
my $job_uuid = $res->getHeader("Job-UUID");

I use same code but in ManagedELS. But, I always get uuid and job_uuid same
value.

My question is, if it is same value then is it necessary to get job_uuid
from res?

Thanks.

Malay Thakershi
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From avi at avimarcus.net  Wed Apr 18 03:54:22 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 18 Apr 2012 02:54:22 +0300
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
Message-ID: <CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>

Are you using the same bgapi or just a plain API?
-Avi


On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com>wrote:

> Hello,
>
> Here is a code snippet from ESL example in wiki.
> http://wiki.freeswitch.org/wiki/Esl#Examples
>
> my $uuid = $con->api("create_uuid")->getBody();
> my $res =
>     $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
> my $job_uuid = $res->getHeader("Job-UUID");
>
> I use same code but in ManagedELS. But, I always get uuid and job_uuid
> same value.
>
> My question is, if it is same value then is it necessary to get job_uuid
> from res?
>
> Thanks.
>
> Malay Thakershi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Wed Apr 18 04:16:09 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 17 Apr 2012 19:16:09 -0500
Subject: [Freeswitch-users] Call transfer failed when bypass media
In-Reply-To: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>
References: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>
Message-ID: <CAKbxfG9LBeBQNhtr5s-DMNt+-2s+=G-0k997Vqii2ugvhESVng@mail.gmail.com>

try the following 2 params in your sofia profile:

<param name="media-option" value="resume-media-on-hold"/>
<param name="media-option" value="bypass-media-after-att-xfer"/>

and use latest GIT

Please do not direct questions to me directly, we have several
official methods of getting FS support and I cannot keep track of
random individual email threads.



On Tue, Apr 17, 2012 at 8:23 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> Dear AM,
>
> i would like to make media bypass for lower FS traffic, however, if i enable
> media bypass, the call transfer failed, the parameters as following,
>
> ?? <param name="inbound-bypass-media" value="true"/>
> ? ? <param name="inbound-proxy-media" value="false"/>
>
> but if i disable media bypass like configuration below, call transfer works
> well.
>
> ? ?<param name="inbound-bypass-media" value="false"/>
> ? ? <param name="inbound-proxy-media" value="false"/>
>
> the?scenario?is very simple, PSTN user X->media GW->FS->Registered user A, A
> answer the call, then press hold button and dial B, A talked with B, A
> hangup the call, pstn user X should talk with B.
>
> Could you please advise if anything else i missed, i below FS so strong, it
> must support this?scenario, i was suffered for this issue for over two
> years... thanks in advance.
>
> --
> Regards,
> Charles
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From rajesh.mbm at gmail.com  Wed Apr 18 08:18:06 2012
From: rajesh.mbm at gmail.com (Rajesh Choudhary)
Date: Wed, 18 Apr 2012 09:48:06 +0530
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
	<CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
Message-ID: <CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>

Thanks for Quick reply
files are not there in  /opt/freeswitch/conf
Following is the output of sudo lsof | grep freeswitch

Thanks,
Rajesh


freeswitc 1097       root  txt       REG        8,6       37464     557911
/usr/local/freeswitch/bin/freeswitch
freeswitc 1097       root  mem       REG        8,6       56818     558014
/usr/local/freeswitch/mod/mod_logfile.so
freeswitc 1097       root  mem       REG        8,6       70488     558085
/usr/local/freeswitch/mod/mod_xml_cdr.so
freeswitc 1097       root  mem       REG        8,6       78418     558016
/usr/local/freeswitch/mod/mod_loopback.so
freeswitc 1097       root  mem       REG        8,6      123376     557992
/usr/local/freeswitch/mod/mod_expr.so
freeswitc 1097       root  mem       REG        8,6       48335     557988
/usr/local/freeswitch/mod/mod_esf.so
freeswitc 1097       root  mem       REG        8,6       60471     557998
/usr/local/freeswitch/mod/mod_fsv.so
freeswitc 1097       root  mem       REG        8,6       65151     557980
/usr/local/freeswitch/mod/mod_dialplan_xml.so
freeswitc 1097       root  mem       REG        8,6     7010899     557907
/usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
freeswitc 1097       root  mem       REG        8,6      234583     557968
/usr/local/freeswitch/mod/mod_commands.so
freeswitc 1097       root  mem       REG        8,6      297534     557970
/usr/local/freeswitch/mod/mod_conference.so
freeswitc 1097       root  mem       REG        8,6       54502     557978
/usr/local/freeswitch/mod/mod_dialplan_asterisk.so
freeswitc 1097       root  mem       REG        8,6      199102     557994
/usr/local/freeswitch/mod/mod_fifo.so
freeswitc 1097       root  mem       REG        8,6      255197     558083
/usr/local/freeswitch/mod/mod_voicemail.so
freeswitc 1097       root  mem       REG        8,6       54099     557966
/usr/local/freeswitch/mod/mod_cluechoo.so
freeswitc 1097       root  mem       REG        8,6      146786     558008
/usr/local/freeswitch/mod/mod_httapi.so
freeswitc 1097       root  mem       REG        8,6       41108     558000
/usr/local/freeswitch/mod/mod_g723_1.so
freeswitc 1097       root  mem       REG        8,6       40884     558002
/usr/local/freeswitch/mod/mod_g729.so
freeswitc 1097       root  mem       REG        8,6       40590     557958
/usr/local/freeswitch/mod/mod_amr.so
freeswitc 1097       root  mem       REG        8,6      170072     558010
/usr/local/freeswitch/mod/mod_ilbc.so
freeswitc 1097       root  mem       REG        8,6       45814     558020
/usr/local/freeswitch/mod/mod_native_file.so
freeswitc 1097       root  mem       REG        8,6       46476     558042
/usr/local/freeswitch/mod/mod_shell_stream.so
freeswitc 1097       root  mem       REG        8,6      405014     557982
/usr/local/freeswitch/mod/mod_dingaling.so
freeswitc 1097       root  mem       REG        8,6      133229     557990
/usr/local/freeswitch/mod/mod_event_socket.so
freeswitc 1097       root  mem       REG        8,6      278775     558056
/usr/local/freeswitch/mod/mod_speex.so
freeswitc 1097       root  mem       REG        8,6      141494     558046
/usr/local/freeswitch/mod/mod_siren.so
freeswitc 1097       root  mem       REG        8,6       70598     558012
/usr/local/freeswitch/mod/mod_local_stream.so
freeswitc 1097       root  mem       REG        8,6       47575     558077
/usr/local/freeswitch/mod/mod_tone_stream.so
freeswitc 1097       root  mem       REG        8,6       83162     558071
/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
freeswitc 1097       root  mem       REG        8,6       58147     558069
/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so
freeswitc 1097       root  mem       REG        8,6       55510     558067
/usr/local/freeswitch/mod/mod_spidermonkey_socket.so
freeswitc 1097       root  mem       REG        8,6       49431     558079
/usr/local/freeswitch/mod/mod_tts_commandline.so
freeswitc 1097       root  mem       REG        8,6       62820     558024
/usr/local/freeswitch/mod/mod_say_en.so
freeswitc 1097       root  mem       REG        8,6       60785     558040
/usr/local/freeswitch/mod/mod_say_zh.so
freeswitc 1097       root  mem       REG        8,6       53090     557972
/usr/local/freeswitch/mod/mod_console.so
freeswitc 1097       root  mem       REG        8,6       93667     558036
/usr/local/freeswitch/mod/mod_say_ru.so
freeswitc 1097       root  mem       REG        8,6       60752     558022
/usr/local/freeswitch/mod/mod_say_de.so
freeswitc 1097       root  mem       REG        8,6       60680     558026
/usr/local/freeswitch/mod/mod_say_es.so
freeswitc 1097       root  mem       REG        8,6       61467     558028
/usr/local/freeswitch/mod/mod_say_fr.so
freeswitc 1097       root  mem       REG        8,6       60021     558032
/usr/local/freeswitch/mod/mod_say_it.so
freeswitc 1097       root  mem       REG        8,6       59940     558034
/usr/local/freeswitch/mod/mod_say_nl.so
freeswitc 1097       root  mem       REG        8,6       55911     558030
/usr/local/freeswitch/mod/mod_say_hu.so
freeswitc 1097       root  mem       REG        8,6     1675414     558062
/usr/local/freeswitch/mod/mod_spidermonkey.so
freeswitc 1097       root  mem       REG        8,6       61373     558038
/usr/local/freeswitch/mod/mod_say_th.so
freeswitc 1097       root  mem       REG        8,6       45694     558004
/usr/local/freeswitch/mod/mod_h26x.so
freeswitc 1097       root  mem       REG        8,6      561707     558475
/usr/local/freeswitch/lib/libnspr4.so
freeswitc 1097       root  mem       REG        8,6      638757     557986
/usr/local/freeswitch/mod/mod_enum.so
freeswitc 1097       root  mem       REG        8,6      215301     557984
/usr/local/freeswitch/mod/mod_dptools.so
freeswitc 1097       root  mem       REG        8,6     1193409     558050
/usr/local/freeswitch/mod/mod_sndfile.so
freeswitc 1097       root  mem       REG        8,6      857708     558018
/usr/local/freeswitch/mod/mod_lua.so
freeswitc 1097       root  mem       REG        8,6       73991     558081
/usr/local/freeswitch/mod/mod_valet_parking.so
freeswitc 1097       root  mem       REG        8,6      196948     558006
/usr/local/freeswitch/mod/mod_hash.so
freeswitc 1097       root  mem       REG        8,6       63432     557976
/usr/local/freeswitch/mod/mod_db.so
freeswitc 1097       root  mem       REG        8,6     1605838     558044
/usr/local/freeswitch/mod/mod_shout.so
freeswitc 1097       root  mem       REG        8,6       62051     557962
/usr/local/freeswitch/mod/mod_cdr_csv.so
freeswitc 1097       root  mem       REG        8,6     5433620     558052
/usr/local/freeswitch/mod/mod_sofia.so
freeswitc 1097       root  mem       REG        8,6      814967     558058
/usr/local/freeswitch/lib/libjs.so.1.0.6
freeswitc 1097       root  mem       REG        8,6     2372252     558054
/usr/local/freeswitch/mod/mod_spandsp.so
freeswitc 1097       root  mem       REG        8,6    19379720     557996
/usr/local/freeswitch/mod/mod_flite.so
freeswitc 1097       root    3wW     REG        8,6           4     525283
/usr/local/freeswitch/run/freeswitch.pid
freeswitc 1097       root    5u      REG        8,6      210944     559634
/usr/local/freeswitch/db/core.db
freeswitc 1097       root    6u      REG        8,6      210944     559634
/usr/local/freeswitch/db/core.db
freeswitc 1097       root    8u      REG        8,6     1709582     559635
/usr/local/freeswitch/log/freeswitch.log
freeswitc 1097       root   12u      REG        8,6       97280     559637
/usr/local/freeswitch/db/sofia_reg_external.db
freeswitc 1097       root   16u      REG        8,6       97280     559638
/usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
freeswitc 1097       root   20u      REG        8,6       97280     559640
/usr/local/freeswitch/db/sofia_reg_internal.db
freeswitc 1097       root   37u      REG        8,6       14336     559639
/usr/local/freeswitch/db/call_limit.db
freeswitc 1097       root   38u      REG        8,6        5120     559641
/usr/local/freeswitch/db/fifo.db
freeswitc 1097       root   39u      REG        8,6       16384     559642
/usr/local/freeswitch/db/voicemail_default.db
freeswitc 1097       root   41r      DIR        8,6        4096     559622
/usr/local/freeswitch/sounds/music/8000
freeswitc 1097       root   43r      DIR        8,6        4096     559622
/usr/local/freeswitch/sounds/music/8000
freeswitc 1097       root   47r      REG        8,6     6104850     559626
/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav
freeswitc 1097       root   48r      REG        8,6     4207128     559623
/usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav
fs_cli    1838       root  txt       REG        8,6      506391     557912
/usr/local/freeswitch/bin/fs_cli
root at rajesh-desktop:~#


On Wed, Apr 18, 2012 at 1:49 AM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>
> wrote:
> > I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04
> LTS
> > as per installation guide, but there are no conf files in
> > /usr/local/freeswitch/conf folder.
>
> They might be in /opt/freeswitch/conf/
>
>
> > However when  i run sofia profile external
> > restart reloadxml in FS cli itruns successfully from where it is getting
> > conf files kindly help
>
> While FreeSWITCH is up, run this command: 'sudo lsof | grep
> freeswitch' and see what files it has open.
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sharad at coraltele.com  Wed Apr 18 08:51:49 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Wed, 18 Apr 2012 10:21:49 +0530
Subject: [Freeswitch-users] Freeswitch and network bonding
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org><4D4D700662D8429188A8259DD92622D3@sharad><CAM21H-Mkkbsjo9iW8Jg4-nHxuJWuCbLYECNc3800BWXkarWoTg@mail.gmail.com><980C368D72A94CC0978872726DA09E7E@sharad>
	<CANBhyPCO6zKQ2UsC2RCYHaR3gdev7jmx0iDiX8Z0k5rmeGxV1g@mail.gmail.com>
Message-ID: <1ACF3A6C0670455C9450E20F29FC100D@sharad>

Hi All 

I am trying the FS with network bonding. My configuration is as follows -

eth0    -    No IP assigned
eth1    -    No IP assigned
bond0  -   192.168.4.65          This is the virtual IP which is to access the freeswitch.

Earlier when I was throwing the call to FS on 192.168.4.65, call was not arriving at FS. But later on with the help of some kind people of this forum, I got some suggestions & I defined the "domain=192.168.4.65" & in internal profile, I have defined sip-ip=domain & rtp-ip=domain. After this, FS is attending the call. 

Now the problem is voice is one way. From FS to Network, voice is OK but Network to FS there is no voice. 

I request the forum to help me on this issue also.

Best Regards
Sharad
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From mthakershi at gmail.com  Wed Apr 18 08:54:37 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Tue, 17 Apr 2012 23:54:37 -0500
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
	<CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>
Message-ID: <CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>

I am using exact same code as given here:
http://wiki.freeswitch.org/wiki/Esl#Examples

bgapi to invoke "originate"

Thanks.

Malay

On Tue, Apr 17, 2012 at 6:54 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Are you using the same bgapi or just a plain API?
> -Avi
>
>
> On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com>wrote:
>
>> Hello,
>>
>> Here is a code snippet from ESL example in wiki.
>> http://wiki.freeswitch.org/wiki/Esl#Examples
>>
>>  my $uuid = $con->api("create_uuid")->getBody();
>> my $res =
>>     $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
>> my $job_uuid = $res->getHeader("Job-UUID");
>>
>> I use same code but in ManagedELS. But, I always get uuid and job_uuid
>> same value.
>>
>> My question is, if it is same value then is it necessary to get job_uuid
>> from res?
>>
>> Thanks.
>>
>> Malay Thakershi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Wed Apr 18 09:11:15 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 18 Apr 2012 09:11:15 +0400
Subject: [Freeswitch-users] SIP trace specific call or uuid
In-Reply-To: <BLU0-SMTP360E3CB3025065987FF9CA3BD3F0@phx.gbl>
References: <CAGX+XzsboC5mUU_SzVtc3BMDRpGJiC7haaGRgNL-6R=mRp0jYw@mail.gmail.com>
	<CALNZuwmW7Y1yNm7uQDUzV1ZVGxm5EjUsByQ86azx1vrwcSjReQ@mail.gmail.com>
	<4F8D74CD.3040606@communicatefreely.net>
	<BLU0-SMTP360E3CB3025065987FF9CA3BD3F0@phx.gbl>
Message-ID: <CANBhyPCfTn_FtcAvQ6h6a1jHKY7E6P3+=HeczcQD6PKpDjtkpA@mail.gmail.com>

It's also useful for debugging some fax issues especially T.38

ngrep -qt -W byline -i t38

17 ?????? 2012 ?. 17:56 ???????????? Barnaby Ritchley <
barnyritchley at hotmail.com> ???????:

> Another option if you just want to view sip messaging is to use ngrep.
>
> ngrep is useful for debugging certain calls as you can filter, ie if you
> wanted to view calls to and from 0123456789 you can do the following:
>
> ngrep -qtW byline 12345678 port 5060
>
> you can replace the phone number with ip addresses also if you wish
>
>
>
> On 17 Apr 2012, at 14:49, Tim St. Pierre wrote:
>
> Yes, I wish there was a way too.
>
> In the interim, I use sofia status profile user XXX to find out their IP
> and port, then on the same machine, I use tcpdump to capture the data.
>
> tcpdump -i INTERFACE -s 1500 -A host IPADDRESS and port SIPPORT will get
> you what you want.
>
> You can also use tcpdump -i INTERFACE -s 1500 -w my_capture.pcap host
> IPADDRESS and port SIPPORT to generate a file that can be opened in
> wireshark.  This is much easier to work with.
>
> Even if you capture everything on the interface, wireshark can sort out
> the individual calls, so it can still be useful.
>
> -Tim
>
> Avi Marcus wrote:
> > Not within FS, yet.
> > Try ngrep - http://wiki.freeswitch.org/wiki/Packet_Capture#ngrep
> >
> > -Avi
> >
> >
> >
> > On Fri, Mar 30, 2012 at 11:59 AM, Anton VG <anton.vazir at gmail.com
> > <mailto:anton.vazir at gmail.com>> wrote:
> >
> >    HI!
> >
> >    Is there a way to enable sip-trace on the specific call or uuid?
> >
> >    I've been only able to find how to enable it per profile, but on a
> >    busy situation, the information amount
> >    is too high to be useful, so it's often necessary to trace a specific
> >    call or call to specific destination.
> >
> >
>  _________________________________________________________________________
> >    Professional FreeSWITCH Consulting Services:
> >    consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >    http://www.freeswitchsolutions.com
> >
> >    
> >    
> >
> >    Official FreeSWITCH Sites
> >    http://www.freeswitch.org
> >    http://wiki.freeswitch.org
> >    http://www.cluecon.com
> >
> >    FreeSWITCH-users mailing list
> >    FreeSWITCH-users at lists.freeswitch.org
> >    <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >    http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Wed Apr 18 09:13:16 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 18 Apr 2012 09:13:16 +0400
Subject: [Freeswitch-users] how to see which extensions are defined?
In-Reply-To: <CAAtWKeHhCfXsPh6uaZEs57Mv9RSj42ZPcj=ni-uq3_F43bOfRw@mail.gmail.com>
References: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
	<CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>
	<CAAtWKeHhCfXsPh6uaZEs57Mv9RSj42ZPcj=ni-uq3_F43bOfRw@mail.gmail.com>
Message-ID: <CANBhyPDFrz+6ZErHVtkX_KJnc3i7k6dpZXbQJyeb2GL+CiP6oA@mail.gmail.com>

root at ippbx:/usr/local/freeswitch/conf# tree directory/

17 ?????? 2012 ?. 22:16 ???????????? Yehavi Bourvine <
yehavi.bourvine at gmail.com> ???????:

> FS does not complain, and the extension file is in the directory file...
>
>                   Thanks, __Yehavi:
>
>
> 2012/4/17 Avi Marcus <avi at avimarcus.net>
>
>> If you include all your config files, then FS will complain about xml
>> parse errors if there are any.
>> So how about you try a simple grep of the config "directory" directory
>> for the extension -- does that help at all?
>>
>> -Avi
>>
>>
>> On Tue, Apr 17, 2012 at 6:22 PM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>> Hello,
>>>
>>>   We have the following odd thing: I've tried updating FreeSwitch
>>> version on my test machine two days ago and since then no extension can be
>>> registered: they all get "403 fobidden". Going back to an older version
>>> makes it work.
>>>
>>> I am missing some tool/command to see which extensions are defined (i.e.
>>> something similar to "sofia status profile XXX reg", but for defined, not
>>> only registered extensions). I would like to make sure that it reads
>>> correctly the extensions definitions.
>>>
>>> Any such thing?
>>>
>>> I do not open JIRA yet as I am not sure whether the problem is with our
>>> config or not.
>>>
>>>                        Thanks, __Yehavi:
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mitch.capper at gmail.com  Wed Apr 18 09:17:42 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Tue, 17 Apr 2012 22:17:42 -0700
Subject: [Freeswitch-users] how to see which extensions are defined?
In-Reply-To: <CANBhyPDFrz+6ZErHVtkX_KJnc3i7k6dpZXbQJyeb2GL+CiP6oA@mail.gmail.com>
References: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
	<CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>
	<CAAtWKeHhCfXsPh6uaZEs57Mv9RSj42ZPcj=ni-uq3_F43bOfRw@mail.gmail.com>
	<CANBhyPDFrz+6ZErHVtkX_KJnc3i7k6dpZXbQJyeb2GL+CiP6oA@mail.gmail.com>
Message-ID: <CALn233rRNRKKy8Uddf-vZ-0fs=i=QR3DYSRRPk4zc9RW9TQtfg@mail.gmail.com>

To test FS parsed the user configs properly you can try and pull a
variable from them and see if that works. There was an API command
/dptools I believe that was added in the last few months to allow you
to do so.

~mitch

2012/4/17 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> root at ippbx:/usr/local/freeswitch/conf# tree directory/
>
> 17 ?????? 2012??. 22:16 ???????????? Yehavi Bourvine
> <yehavi.bourvine at gmail.com> ???????:
>
>> FS does not complain, and the extension file is in the directory file...
>>
>> ????????????????? Thanks, __Yehavi:
>>
>>
>> 2012/4/17 Avi Marcus <avi at avimarcus.net>
>>>
>>> If you include all your config files, then FS will complain about xml
>>> parse errors if there are any.
>>> So how about you try a simple grep of the config "directory" directory
>>> for the extension -- does that help at all?
>>>
>>> -Avi
>>>
>>>
>>> On Tue, Apr 17, 2012 at 6:22 PM, Yehavi Bourvine
>>> <yehavi.bourvine at gmail.com> wrote:
>>>>
>>>> Hello,
>>>>
>>>> ? We have the following odd thing: I've tried updating FreeSwitch
>>>> version on my test machine two days ago and since then no extension can be
>>>> registered: they all get "403 fobidden". Going back to an older version
>>>> makes it work.
>>>>
>>>> I am missing some tool/command to see which extensions are defined (i.e.
>>>> something similar to "sofia status profile XXX reg", but for defined, not
>>>> only registered extensions). I would like to make sure that it reads
>>>> correctly the extensions definitions.
>>>>
>>>> Any such thing?
>>>>
>>>> I do not open JIRA yet as I am not sure whether the problem is with our
>>>> config or not.
>>>>
>>>> ?????????????????????? Thanks, __Yehavi:
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Wed Apr 18 09:43:18 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 18 Apr 2012 05:43:18 +0000
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
	<CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>,
	<CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>
Message-ID: <CE89AF39-66C6-4140-AFB8-28CEBA8A6F83@visionutveckling.se>

Seems strange. These must be two different values. The first uuid is generated for the actual call, the second is a uuid to help you keep track of the background job. Are you sure they are the same, and not just very similar?

/Peter

18 apr 2012 kl. 06:58 skrev "Malay Thakershi" <mthakershi at gmail.com<mailto:mthakershi at gmail.com>>:

I am using exact same code as given here:  http://wiki.freeswitch.org/wiki/Esl#Examples

bgapi to invoke "originate"

Thanks.

Malay

On Tue, Apr 17, 2012 at 6:54 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
Are you using the same bgapi or just a plain API?
-Avi


On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com<mailto:mthakershi at gmail.com>> wrote:
Hello,

Here is a code snippet from ESL example in wiki. http://wiki.freeswitch.org/wiki/Esl#Examples

my $uuid = $con->api("create_uuid")->getBody();
my $res =
    $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
my $job_uuid = $res->getHeader("Job-UUID");

I use same code but in ManagedELS. But, I always get uuid and job_uuid same value.

My question is, if it is same value then is it necessary to get job_uuid from res?

Thanks.

Malay Thakershi

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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From fieldpeak at gmail.com  Wed Apr 18 10:13:24 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 18 Apr 2012 14:13:24 +0800
Subject: [Freeswitch-users] Call transfer failed when bypass media
In-Reply-To: <CAKbxfG9LBeBQNhtr5s-DMNt+-2s+=G-0k997Vqii2ugvhESVng@mail.gmail.com>
References: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>
	<CAKbxfG9LBeBQNhtr5s-DMNt+-2s+=G-0k997Vqii2ugvhESVng@mail.gmail.com>
Message-ID: <CAOTmbMyUkTjSLH2MW79j2-2E1wNkQsTYYGoR2=5WTsshKZVJuQ@mail.gmail.com>

AM,

Thank you very much! i'll try them.
i ever saw these two parameters in wiki, however, i understand it the value
of "media-option" can be either "resume-media-on-hold" or
"bypass-media-after-att-xfer", can not configure both...so i did not try
both yet...

usually i send support request to FreeSWITCH-users at lists.freeswitch.org,
however, most of them has no any response, could you please advise more
efficient way, so far i know only this way for requesting support, thanks a
lot!

Regards,
Charles

2012/4/18 Anthony Minessale <anthony.minessale at gmail.com>

> try the following 2 params in your sofia profile:
>
> <param name="media-option" value="resume-media-on-hold"/>
> <param name="media-option" value="bypass-media-after-att-xfer"/>
>
> and use latest GIT
>
> Please do not direct questions to me directly, we have several
> official methods of getting FS support and I cannot keep track of
> random individual email threads.
>
>
>
> On Tue, Apr 17, 2012 at 8:23 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> > Dear AM,
> >
> > i would like to make media bypass for lower FS traffic, however, if i
> enable
> > media bypass, the call transfer failed, the parameters as following,
> >
> >    <param name="inbound-bypass-media" value="true"/>
> >     <param name="inbound-proxy-media" value="false"/>
> >
> > but if i disable media bypass like configuration below, call transfer
> works
> > well.
> >
> >    <param name="inbound-bypass-media" value="false"/>
> >     <param name="inbound-proxy-media" value="false"/>
> >
> > the scenario is very simple, PSTN user X->media GW->FS->Registered user
> A, A
> > answer the call, then press hold button and dial B, A talked with B, A
> > hangup the call, pstn user X should talk with B.
> >
> > Could you please advise if anything else i missed, i below FS so strong,
> it
> > must support this scenario, i was suffered for this issue for over two
> > years... thanks in advance.
> >
> > --
> > Regards,
> > Charles
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
Regards,
Charles
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From bernard.david.murphy at gmail.com  Wed Apr 18 03:13:01 2012
From: bernard.david.murphy at gmail.com (Bernard Murphy)
Date: Wed, 18 Apr 2012 00:13:01 +0100
Subject: [Freeswitch-users]  Translating DTMF from RFC2833 to INFO
Message-ID: <CAOte9J571o3A13CsBUVx9SAbK+HTrCoTk+pZfZEz+6Rt4aU=HQ@mail.gmail.com>

I appreciate this message refers to a very old post, but I have EXACTLY the
same problem and would love to know if you got this fixed in the end and
how ?

Basically Cisco gateway not sending DTMF events to the ISDN when they are
received by RFC2833 however do get sent when received by SIP INFO.

[original post]
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-December/050780.html

Thanks
Bernie
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From bdfoster at endigotech.com  Wed Apr 18 10:38:31 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 18 Apr 2012 02:38:31 -0400
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
	<CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
	<CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>
Message-ID: <CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>

Try:
sudo lsof | grep default

-BDF


On Wed, Apr 18, 2012 at 12:18 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>wrote:

> Thanks for Quick reply
> files are not there in  /opt/freeswitch/conf
> Following is the output of sudo lsof | grep freeswitch
>
> Thanks,
> Rajesh
>
>
> freeswitc 1097       root  txt       REG        8,6       37464     557911
> /usr/local/freeswitch/bin/freeswitch
> freeswitc 1097       root  mem       REG        8,6       56818     558014
> /usr/local/freeswitch/mod/mod_logfile.so
> freeswitc 1097       root  mem       REG        8,6       70488     558085
> /usr/local/freeswitch/mod/mod_xml_cdr.so
> freeswitc 1097       root  mem       REG        8,6       78418     558016
> /usr/local/freeswitch/mod/mod_loopback.so
> freeswitc 1097       root  mem       REG        8,6      123376     557992
> /usr/local/freeswitch/mod/mod_expr.so
> freeswitc 1097       root  mem       REG        8,6       48335     557988
> /usr/local/freeswitch/mod/mod_esf.so
> freeswitc 1097       root  mem       REG        8,6       60471     557998
> /usr/local/freeswitch/mod/mod_fsv.so
> freeswitc 1097       root  mem       REG        8,6       65151     557980
> /usr/local/freeswitch/mod/mod_dialplan_xml.so
> freeswitc 1097       root  mem       REG        8,6     7010899     557907
> /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
> freeswitc 1097       root  mem       REG        8,6      234583     557968
> /usr/local/freeswitch/mod/mod_commands.so
> freeswitc 1097       root  mem       REG        8,6      297534     557970
> /usr/local/freeswitch/mod/mod_conference.so
> freeswitc 1097       root  mem       REG        8,6       54502     557978
> /usr/local/freeswitch/mod/mod_dialplan_asterisk.so
> freeswitc 1097       root  mem       REG        8,6      199102     557994
> /usr/local/freeswitch/mod/mod_fifo.so
> freeswitc 1097       root  mem       REG        8,6      255197     558083
> /usr/local/freeswitch/mod/mod_voicemail.so
> freeswitc 1097       root  mem       REG        8,6       54099     557966
> /usr/local/freeswitch/mod/mod_cluechoo.so
> freeswitc 1097       root  mem       REG        8,6      146786     558008
> /usr/local/freeswitch/mod/mod_httapi.so
> freeswitc 1097       root  mem       REG        8,6       41108     558000
> /usr/local/freeswitch/mod/mod_g723_1.so
> freeswitc 1097       root  mem       REG        8,6       40884     558002
> /usr/local/freeswitch/mod/mod_g729.so
> freeswitc 1097       root  mem       REG        8,6       40590     557958
> /usr/local/freeswitch/mod/mod_amr.so
> freeswitc 1097       root  mem       REG        8,6      170072     558010
> /usr/local/freeswitch/mod/mod_ilbc.so
> freeswitc 1097       root  mem       REG        8,6       45814     558020
> /usr/local/freeswitch/mod/mod_native_file.so
> freeswitc 1097       root  mem       REG        8,6       46476     558042
> /usr/local/freeswitch/mod/mod_shell_stream.so
> freeswitc 1097       root  mem       REG        8,6      405014     557982
> /usr/local/freeswitch/mod/mod_dingaling.so
> freeswitc 1097       root  mem       REG        8,6      133229     557990
> /usr/local/freeswitch/mod/mod_event_socket.so
> freeswitc 1097       root  mem       REG        8,6      278775     558056
> /usr/local/freeswitch/mod/mod_speex.so
> freeswitc 1097       root  mem       REG        8,6      141494     558046
> /usr/local/freeswitch/mod/mod_siren.so
> freeswitc 1097       root  mem       REG        8,6       70598     558012
> /usr/local/freeswitch/mod/mod_local_stream.so
> freeswitc 1097       root  mem       REG        8,6       47575     558077
> /usr/local/freeswitch/mod/mod_tone_stream.so
> freeswitc 1097       root  mem       REG        8,6       83162     558071
> /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
> freeswitc 1097       root  mem       REG        8,6       58147     558069
> /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so
> freeswitc 1097       root  mem       REG        8,6       55510     558067
> /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
> freeswitc 1097       root  mem       REG        8,6       49431     558079
> /usr/local/freeswitch/mod/mod_tts_commandline.so
> freeswitc 1097       root  mem       REG        8,6       62820     558024
> /usr/local/freeswitch/mod/mod_say_en.so
> freeswitc 1097       root  mem       REG        8,6       60785     558040
> /usr/local/freeswitch/mod/mod_say_zh.so
> freeswitc 1097       root  mem       REG        8,6       53090     557972
> /usr/local/freeswitch/mod/mod_console.so
> freeswitc 1097       root  mem       REG        8,6       93667     558036
> /usr/local/freeswitch/mod/mod_say_ru.so
> freeswitc 1097       root  mem       REG        8,6       60752     558022
> /usr/local/freeswitch/mod/mod_say_de.so
> freeswitc 1097       root  mem       REG        8,6       60680     558026
> /usr/local/freeswitch/mod/mod_say_es.so
> freeswitc 1097       root  mem       REG        8,6       61467     558028
> /usr/local/freeswitch/mod/mod_say_fr.so
> freeswitc 1097       root  mem       REG        8,6       60021     558032
> /usr/local/freeswitch/mod/mod_say_it.so
> freeswitc 1097       root  mem       REG        8,6       59940     558034
> /usr/local/freeswitch/mod/mod_say_nl.so
> freeswitc 1097       root  mem       REG        8,6       55911     558030
> /usr/local/freeswitch/mod/mod_say_hu.so
> freeswitc 1097       root  mem       REG        8,6     1675414     558062
> /usr/local/freeswitch/mod/mod_spidermonkey.so
> freeswitc 1097       root  mem       REG        8,6       61373     558038
> /usr/local/freeswitch/mod/mod_say_th.so
> freeswitc 1097       root  mem       REG        8,6       45694     558004
> /usr/local/freeswitch/mod/mod_h26x.so
> freeswitc 1097       root  mem       REG        8,6      561707     558475
> /usr/local/freeswitch/lib/libnspr4.so
> freeswitc 1097       root  mem       REG        8,6      638757     557986
> /usr/local/freeswitch/mod/mod_enum.so
> freeswitc 1097       root  mem       REG        8,6      215301     557984
> /usr/local/freeswitch/mod/mod_dptools.so
> freeswitc 1097       root  mem       REG        8,6     1193409     558050
> /usr/local/freeswitch/mod/mod_sndfile.so
> freeswitc 1097       root  mem       REG        8,6      857708     558018
> /usr/local/freeswitch/mod/mod_lua.so
> freeswitc 1097       root  mem       REG        8,6       73991     558081
> /usr/local/freeswitch/mod/mod_valet_parking.so
> freeswitc 1097       root  mem       REG        8,6      196948     558006
> /usr/local/freeswitch/mod/mod_hash.so
> freeswitc 1097       root  mem       REG        8,6       63432     557976
> /usr/local/freeswitch/mod/mod_db.so
> freeswitc 1097       root  mem       REG        8,6     1605838     558044
> /usr/local/freeswitch/mod/mod_shout.so
> freeswitc 1097       root  mem       REG        8,6       62051     557962
> /usr/local/freeswitch/mod/mod_cdr_csv.so
> freeswitc 1097       root  mem       REG        8,6     5433620     558052
> /usr/local/freeswitch/mod/mod_sofia.so
> freeswitc 1097       root  mem       REG        8,6      814967     558058
> /usr/local/freeswitch/lib/libjs.so.1.0.6
> freeswitc 1097       root  mem       REG        8,6     2372252     558054
> /usr/local/freeswitch/mod/mod_spandsp.so
> freeswitc 1097       root  mem       REG        8,6    19379720     557996
> /usr/local/freeswitch/mod/mod_flite.so
> freeswitc 1097       root    3wW     REG        8,6           4     525283
> /usr/local/freeswitch/run/freeswitch.pid
> freeswitc 1097       root    5u      REG        8,6      210944     559634
> /usr/local/freeswitch/db/core.db
> freeswitc 1097       root    6u      REG        8,6      210944     559634
> /usr/local/freeswitch/db/core.db
> freeswitc 1097       root    8u      REG        8,6     1709582     559635
> /usr/local/freeswitch/log/freeswitch.log
> freeswitc 1097       root   12u      REG        8,6       97280     559637
> /usr/local/freeswitch/db/sofia_reg_external.db
> freeswitc 1097       root   16u      REG        8,6       97280     559638
> /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
> freeswitc 1097       root   20u      REG        8,6       97280     559640
> /usr/local/freeswitch/db/sofia_reg_internal.db
> freeswitc 1097       root   37u      REG        8,6       14336     559639
> /usr/local/freeswitch/db/call_limit.db
> freeswitc 1097       root   38u      REG        8,6        5120     559641
> /usr/local/freeswitch/db/fifo.db
> freeswitc 1097       root   39u      REG        8,6       16384     559642
> /usr/local/freeswitch/db/voicemail_default.db
> freeswitc 1097       root   41r      DIR        8,6        4096     559622
> /usr/local/freeswitch/sounds/music/8000
> freeswitc 1097       root   43r      DIR        8,6        4096     559622
> /usr/local/freeswitch/sounds/music/8000
> freeswitc 1097       root   47r      REG        8,6     6104850     559626
> /usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav
> freeswitc 1097       root   48r      REG        8,6     4207128     559623
> /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav
> fs_cli    1838       root  txt       REG        8,6      506391     557912
> /usr/local/freeswitch/bin/fs_cli
> root at rajesh-desktop:~#
>
>
> On Wed, Apr 18, 2012 at 1:49 AM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>
>> wrote:
>> > I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04
>> LTS
>> > as per installation guide, but there are no conf files in
>> > /usr/local/freeswitch/conf folder.
>>
>> They might be in /opt/freeswitch/conf/
>>
>>
>> > However when  i run sofia profile external
>> > restart reloadxml in FS cli itruns successfully from where it is getting
>> > conf files kindly help
>>
>> While FreeSWITCH is up, run this command: 'sudo lsof | grep
>> freeswitch' and see what files it has open.
>>
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
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From bdfoster at endigotech.com  Wed Apr 18 10:40:11 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 18 Apr 2012 02:40:11 -0400
Subject: [Freeswitch-users] Freeswitch and network bonding
In-Reply-To: <1ACF3A6C0670455C9450E20F29FC100D@sharad>
References: <mailman.7401.1334656091.13443.freeswitch-users@lists.freeswitch.org>
	<4D4D700662D8429188A8259DD92622D3@sharad>
	<CAM21H-Mkkbsjo9iW8Jg4-nHxuJWuCbLYECNc3800BWXkarWoTg@mail.gmail.com>
	<980C368D72A94CC0978872726DA09E7E@sharad>
	<CANBhyPCO6zKQ2UsC2RCYHaR3gdev7jmx0iDiX8Z0k5rmeGxV1g@mail.gmail.com>
	<1ACF3A6C0670455C9450E20F29FC100D@sharad>
Message-ID: <CAKV-jFyTEx-F5Q9A8EVaccdh1nCo6ArBboJHcpk+=5tHfMa_MQ@mail.gmail.com>

sip-ip and rtp-ip should be the ip address not "domain"

-BDF

2012/4/18 Sharad Garg <sharad at coraltele.com>

> **
> **Hi All
>
> I am trying the FS with network bonding. My configuration is as follows -
>
> eth0    -    No IP assigned
> eth1    -    No IP assigned
> bond0  -   192.168.4.65          This is the virtual IP which is to access
> the freeswitch.
> Earlier when I was throwing the call to FS on 192.168.4.65, call was not
> arriving at FS. But later on with the help of some kind people of this
> forum, I got some suggestions & I defined the "domain=192.168.4.65" & in
> internal profile, I have defined sip-ip=domain & rtp-ip=domain. After this,
> FS is attending the call.
>
> Now the problem is voice is one way. >From FS to Network, voice is OK but
> Network to FS there is no voice.
>
> I request the forum to help me on this issue also.
>
> Best Regards
> Sharad
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
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From rajesh.mbm at gmail.com  Wed Apr 18 11:23:07 2012
From: rajesh.mbm at gmail.com (Rajesh Choudhary)
Date: Wed, 18 Apr 2012 12:53:07 +0530
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
	<CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
	<CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>
	<CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>
Message-ID: <CAAMWH7qKJ0H+A+GYc+9stwwrm5-cTkw-27Ch_C+BOs7fpFJ50A@mail.gmail.com>

still confused

freeswitch at internal>sofia profile external restart reloadxml
freeswitch at internal> sofia status
                     Name       Type
                                      Data    State
=================================================================================================
                 external    profile
sip:mod_sofia at 127.0.0.1:5080    RUNNING (0)
    external::example.com    gateway
sip:joeuser at example.com    NOREG
            internal-ipv6    profile
sip:mod_sofia@[::1]:5060
RUNNING (0)
                127.0.0.1      alias
internal    ALIASED
                 internal    profile
sip:mod_sofia at 127.0.0.1:5060    RUNNING (0)
=================================================================================================
3 profiles 1 alias


where is the configuration file for this example.com gateway

help

On Wed, Apr 18, 2012 at 12:08 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Try:
> sudo lsof | grep default
>
> -BDF
>
>
> On Wed, Apr 18, 2012 at 12:18 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>wrote:
>
>> Thanks for Quick reply
>> files are not there in  /opt/freeswitch/conf
>> Following is the output of sudo lsof | grep freeswitch
>>
>> Thanks,
>> Rajesh
>>
>>
>> freeswitc 1097       root  txt       REG        8,6       37464
>> 557911 /usr/local/freeswitch/bin/freeswitch
>> freeswitc 1097       root  mem       REG        8,6       56818
>> 558014 /usr/local/freeswitch/mod/mod_logfile.so
>> freeswitc 1097       root  mem       REG        8,6       70488
>> 558085 /usr/local/freeswitch/mod/mod_xml_cdr.so
>> freeswitc 1097       root  mem       REG        8,6       78418
>> 558016 /usr/local/freeswitch/mod/mod_loopback.so
>> freeswitc 1097       root  mem       REG        8,6      123376
>> 557992 /usr/local/freeswitch/mod/mod_expr.so
>> freeswitc 1097       root  mem       REG        8,6       48335
>> 557988 /usr/local/freeswitch/mod/mod_esf.so
>> freeswitc 1097       root  mem       REG        8,6       60471
>> 557998 /usr/local/freeswitch/mod/mod_fsv.so
>> freeswitc 1097       root  mem       REG        8,6       65151
>> 557980 /usr/local/freeswitch/mod/mod_dialplan_xml.so
>> freeswitc 1097       root  mem       REG        8,6     7010899
>> 557907 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
>> freeswitc 1097       root  mem       REG        8,6      234583
>> 557968 /usr/local/freeswitch/mod/mod_commands.so
>> freeswitc 1097       root  mem       REG        8,6      297534
>> 557970 /usr/local/freeswitch/mod/mod_conference.so
>> freeswitc 1097       root  mem       REG        8,6       54502
>> 557978 /usr/local/freeswitch/mod/mod_dialplan_asterisk.so
>> freeswitc 1097       root  mem       REG        8,6      199102
>> 557994 /usr/local/freeswitch/mod/mod_fifo.so
>> freeswitc 1097       root  mem       REG        8,6      255197
>> 558083 /usr/local/freeswitch/mod/mod_voicemail.so
>> freeswitc 1097       root  mem       REG        8,6       54099
>> 557966 /usr/local/freeswitch/mod/mod_cluechoo.so
>> freeswitc 1097       root  mem       REG        8,6      146786
>> 558008 /usr/local/freeswitch/mod/mod_httapi.so
>> freeswitc 1097       root  mem       REG        8,6       41108
>> 558000 /usr/local/freeswitch/mod/mod_g723_1.so
>> freeswitc 1097       root  mem       REG        8,6       40884
>> 558002 /usr/local/freeswitch/mod/mod_g729.so
>> freeswitc 1097       root  mem       REG        8,6       40590
>> 557958 /usr/local/freeswitch/mod/mod_amr.so
>> freeswitc 1097       root  mem       REG        8,6      170072
>> 558010 /usr/local/freeswitch/mod/mod_ilbc.so
>> freeswitc 1097       root  mem       REG        8,6       45814
>> 558020 /usr/local/freeswitch/mod/mod_native_file.so
>> freeswitc 1097       root  mem       REG        8,6       46476
>> 558042 /usr/local/freeswitch/mod/mod_shell_stream.so
>> freeswitc 1097       root  mem       REG        8,6      405014
>> 557982 /usr/local/freeswitch/mod/mod_dingaling.so
>> freeswitc 1097       root  mem       REG        8,6      133229
>> 557990 /usr/local/freeswitch/mod/mod_event_socket.so
>> freeswitc 1097       root  mem       REG        8,6      278775
>> 558056 /usr/local/freeswitch/mod/mod_speex.so
>> freeswitc 1097       root  mem       REG        8,6      141494
>> 558046 /usr/local/freeswitch/mod/mod_siren.so
>> freeswitc 1097       root  mem       REG        8,6       70598
>> 558012 /usr/local/freeswitch/mod/mod_local_stream.so
>> freeswitc 1097       root  mem       REG        8,6       47575
>> 558077 /usr/local/freeswitch/mod/mod_tone_stream.so
>> freeswitc 1097       root  mem       REG        8,6       83162
>> 558071 /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>> freeswitc 1097       root  mem       REG        8,6       58147
>> 558069 /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so
>> freeswitc 1097       root  mem       REG        8,6       55510
>> 558067 /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>> freeswitc 1097       root  mem       REG        8,6       49431
>> 558079 /usr/local/freeswitch/mod/mod_tts_commandline.so
>> freeswitc 1097       root  mem       REG        8,6       62820
>> 558024 /usr/local/freeswitch/mod/mod_say_en.so
>> freeswitc 1097       root  mem       REG        8,6       60785
>> 558040 /usr/local/freeswitch/mod/mod_say_zh.so
>> freeswitc 1097       root  mem       REG        8,6       53090
>> 557972 /usr/local/freeswitch/mod/mod_console.so
>> freeswitc 1097       root  mem       REG        8,6       93667
>> 558036 /usr/local/freeswitch/mod/mod_say_ru.so
>> freeswitc 1097       root  mem       REG        8,6       60752
>> 558022 /usr/local/freeswitch/mod/mod_say_de.so
>> freeswitc 1097       root  mem       REG        8,6       60680
>> 558026 /usr/local/freeswitch/mod/mod_say_es.so
>> freeswitc 1097       root  mem       REG        8,6       61467
>> 558028 /usr/local/freeswitch/mod/mod_say_fr.so
>> freeswitc 1097       root  mem       REG        8,6       60021
>> 558032 /usr/local/freeswitch/mod/mod_say_it.so
>> freeswitc 1097       root  mem       REG        8,6       59940
>> 558034 /usr/local/freeswitch/mod/mod_say_nl.so
>> freeswitc 1097       root  mem       REG        8,6       55911
>> 558030 /usr/local/freeswitch/mod/mod_say_hu.so
>> freeswitc 1097       root  mem       REG        8,6     1675414
>> 558062 /usr/local/freeswitch/mod/mod_spidermonkey.so
>> freeswitc 1097       root  mem       REG        8,6       61373
>> 558038 /usr/local/freeswitch/mod/mod_say_th.so
>> freeswitc 1097       root  mem       REG        8,6       45694
>> 558004 /usr/local/freeswitch/mod/mod_h26x.so
>> freeswitc 1097       root  mem       REG        8,6      561707
>> 558475 /usr/local/freeswitch/lib/libnspr4.so
>> freeswitc 1097       root  mem       REG        8,6      638757
>> 557986 /usr/local/freeswitch/mod/mod_enum.so
>> freeswitc 1097       root  mem       REG        8,6      215301
>> 557984 /usr/local/freeswitch/mod/mod_dptools.so
>> freeswitc 1097       root  mem       REG        8,6     1193409
>> 558050 /usr/local/freeswitch/mod/mod_sndfile.so
>> freeswitc 1097       root  mem       REG        8,6      857708
>> 558018 /usr/local/freeswitch/mod/mod_lua.so
>> freeswitc 1097       root  mem       REG        8,6       73991
>> 558081 /usr/local/freeswitch/mod/mod_valet_parking.so
>> freeswitc 1097       root  mem       REG        8,6      196948
>> 558006 /usr/local/freeswitch/mod/mod_hash.so
>> freeswitc 1097       root  mem       REG        8,6       63432
>> 557976 /usr/local/freeswitch/mod/mod_db.so
>> freeswitc 1097       root  mem       REG        8,6     1605838
>> 558044 /usr/local/freeswitch/mod/mod_shout.so
>> freeswitc 1097       root  mem       REG        8,6       62051
>> 557962 /usr/local/freeswitch/mod/mod_cdr_csv.so
>> freeswitc 1097       root  mem       REG        8,6     5433620
>> 558052 /usr/local/freeswitch/mod/mod_sofia.so
>> freeswitc 1097       root  mem       REG        8,6      814967
>> 558058 /usr/local/freeswitch/lib/libjs.so.1.0.6
>> freeswitc 1097       root  mem       REG        8,6     2372252
>> 558054 /usr/local/freeswitch/mod/mod_spandsp.so
>> freeswitc 1097       root  mem       REG        8,6    19379720
>> 557996 /usr/local/freeswitch/mod/mod_flite.so
>> freeswitc 1097       root    3wW     REG        8,6           4
>> 525283 /usr/local/freeswitch/run/freeswitch.pid
>> freeswitc 1097       root    5u      REG        8,6      210944
>> 559634 /usr/local/freeswitch/db/core.db
>> freeswitc 1097       root    6u      REG        8,6      210944
>> 559634 /usr/local/freeswitch/db/core.db
>> freeswitc 1097       root    8u      REG        8,6     1709582
>> 559635 /usr/local/freeswitch/log/freeswitch.log
>> freeswitc 1097       root   12u      REG        8,6       97280
>> 559637 /usr/local/freeswitch/db/sofia_reg_external.db
>> freeswitc 1097       root   16u      REG        8,6       97280
>> 559638 /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
>> freeswitc 1097       root   20u      REG        8,6       97280
>> 559640 /usr/local/freeswitch/db/sofia_reg_internal.db
>> freeswitc 1097       root   37u      REG        8,6       14336
>> 559639 /usr/local/freeswitch/db/call_limit.db
>> freeswitc 1097       root   38u      REG        8,6        5120
>> 559641 /usr/local/freeswitch/db/fifo.db
>> freeswitc 1097       root   39u      REG        8,6       16384
>> 559642 /usr/local/freeswitch/db/voicemail_default.db
>> freeswitc 1097       root   41r      DIR        8,6        4096
>> 559622 /usr/local/freeswitch/sounds/music/8000
>> freeswitc 1097       root   43r      DIR        8,6        4096
>> 559622 /usr/local/freeswitch/sounds/music/8000
>> freeswitc 1097       root   47r      REG        8,6     6104850
>> 559626
>> /usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav
>> freeswitc 1097       root   48r      REG        8,6     4207128
>> 559623
>> /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav
>> fs_cli    1838       root  txt       REG        8,6      506391
>> 557912 /usr/local/freeswitch/bin/fs_cli
>> root at rajesh-desktop:~#
>>
>>
>> On Wed, Apr 18, 2012 at 1:49 AM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>
>>> wrote:
>>> > I am a newbie to freeswitch i have installed freeswitch on Ubuntu
>>> 10.04 LTS
>>> > as per installation guide, but there are no conf files in
>>> > /usr/local/freeswitch/conf folder.
>>>
>>> They might be in /opt/freeswitch/conf/
>>>
>>>
>>> > However when  i run sofia profile external
>>> > restart reloadxml in FS cli itruns successfully from where it is
>>> getting
>>> > conf files kindly help
>>>
>>> While FreeSWITCH is up, run this command: 'sudo lsof | grep
>>> freeswitch' and see what files it has open.
>>>
>>>
>>> Best,
>>> Gabe
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Wed Apr 18 11:28:33 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 18 Apr 2012 11:28:33 +0400
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
	<CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
	<CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>
	<CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>
Message-ID: <CANBhyPBS5_PYjmHmc8ns_Vmk-zAX2Q7ZYSspGKOLyniYnSnhCg@mail.gmail.com>

Ok. You can try

strace -e trace=open /usr/local/freeswitch/bin/freeswitch 2>&1 | grep conf

18 ?????? 2012 ?. 10:38 ???????????? Brian Foster
<bdfoster at endigotech.com>???????:

> Try:
> sudo lsof | grep default
>
> -BDF
>
>
>
> On Wed, Apr 18, 2012 at 12:18 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>wrote:
>
>> Thanks for Quick reply
>> files are not there in  /opt/freeswitch/conf
>> Following is the output of sudo lsof | grep freeswitch
>>
>> Thanks,
>> Rajesh
>>
>>
>> freeswitc 1097       root  txt       REG        8,6       37464
>> 557911 /usr/local/freeswitch/bin/freeswitch
>> freeswitc 1097       root  mem       REG        8,6       56818
>> 558014 /usr/local/freeswitch/mod/mod_logfile.so
>> freeswitc 1097       root  mem       REG        8,6       70488
>> 558085 /usr/local/freeswitch/mod/mod_xml_cdr.so
>> freeswitc 1097       root  mem       REG        8,6       78418
>> 558016 /usr/local/freeswitch/mod/mod_loopback.so
>> freeswitc 1097       root  mem       REG        8,6      123376
>> 557992 /usr/local/freeswitch/mod/mod_expr.so
>> freeswitc 1097       root  mem       REG        8,6       48335
>> 557988 /usr/local/freeswitch/mod/mod_esf.so
>> freeswitc 1097       root  mem       REG        8,6       60471
>> 557998 /usr/local/freeswitch/mod/mod_fsv.so
>> freeswitc 1097       root  mem       REG        8,6       65151
>> 557980 /usr/local/freeswitch/mod/mod_dialplan_xml.so
>> freeswitc 1097       root  mem       REG        8,6     7010899
>> 557907 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
>> freeswitc 1097       root  mem       REG        8,6      234583
>> 557968 /usr/local/freeswitch/mod/mod_commands.so
>> freeswitc 1097       root  mem       REG        8,6      297534
>> 557970 /usr/local/freeswitch/mod/mod_conference.so
>> freeswitc 1097       root  mem       REG        8,6       54502
>> 557978 /usr/local/freeswitch/mod/mod_dialplan_asterisk.so
>> freeswitc 1097       root  mem       REG        8,6      199102
>> 557994 /usr/local/freeswitch/mod/mod_fifo.so
>> freeswitc 1097       root  mem       REG        8,6      255197
>> 558083 /usr/local/freeswitch/mod/mod_voicemail.so
>> freeswitc 1097       root  mem       REG        8,6       54099
>> 557966 /usr/local/freeswitch/mod/mod_cluechoo.so
>> freeswitc 1097       root  mem       REG        8,6      146786
>> 558008 /usr/local/freeswitch/mod/mod_httapi.so
>> freeswitc 1097       root  mem       REG        8,6       41108
>> 558000 /usr/local/freeswitch/mod/mod_g723_1.so
>> freeswitc 1097       root  mem       REG        8,6       40884
>> 558002 /usr/local/freeswitch/mod/mod_g729.so
>> freeswitc 1097       root  mem       REG        8,6       40590
>> 557958 /usr/local/freeswitch/mod/mod_amr.so
>> freeswitc 1097       root  mem       REG        8,6      170072
>> 558010 /usr/local/freeswitch/mod/mod_ilbc.so
>> freeswitc 1097       root  mem       REG        8,6       45814
>> 558020 /usr/local/freeswitch/mod/mod_native_file.so
>> freeswitc 1097       root  mem       REG        8,6       46476
>> 558042 /usr/local/freeswitch/mod/mod_shell_stream.so
>> freeswitc 1097       root  mem       REG        8,6      405014
>> 557982 /usr/local/freeswitch/mod/mod_dingaling.so
>> freeswitc 1097       root  mem       REG        8,6      133229
>> 557990 /usr/local/freeswitch/mod/mod_event_socket.so
>> freeswitc 1097       root  mem       REG        8,6      278775
>> 558056 /usr/local/freeswitch/mod/mod_speex.so
>> freeswitc 1097       root  mem       REG        8,6      141494
>> 558046 /usr/local/freeswitch/mod/mod_siren.so
>> freeswitc 1097       root  mem       REG        8,6       70598
>> 558012 /usr/local/freeswitch/mod/mod_local_stream.so
>> freeswitc 1097       root  mem       REG        8,6       47575
>> 558077 /usr/local/freeswitch/mod/mod_tone_stream.so
>> freeswitc 1097       root  mem       REG        8,6       83162
>> 558071 /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>> freeswitc 1097       root  mem       REG        8,6       58147
>> 558069 /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so
>> freeswitc 1097       root  mem       REG        8,6       55510
>> 558067 /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>> freeswitc 1097       root  mem       REG        8,6       49431
>> 558079 /usr/local/freeswitch/mod/mod_tts_commandline.so
>> freeswitc 1097       root  mem       REG        8,6       62820
>> 558024 /usr/local/freeswitch/mod/mod_say_en.so
>> freeswitc 1097       root  mem       REG        8,6       60785
>> 558040 /usr/local/freeswitch/mod/mod_say_zh.so
>> freeswitc 1097       root  mem       REG        8,6       53090
>> 557972 /usr/local/freeswitch/mod/mod_console.so
>> freeswitc 1097       root  mem       REG        8,6       93667
>> 558036 /usr/local/freeswitch/mod/mod_say_ru.so
>> freeswitc 1097       root  mem       REG        8,6       60752
>> 558022 /usr/local/freeswitch/mod/mod_say_de.so
>> freeswitc 1097       root  mem       REG        8,6       60680
>> 558026 /usr/local/freeswitch/mod/mod_say_es.so
>> freeswitc 1097       root  mem       REG        8,6       61467
>> 558028 /usr/local/freeswitch/mod/mod_say_fr.so
>> freeswitc 1097       root  mem       REG        8,6       60021
>> 558032 /usr/local/freeswitch/mod/mod_say_it.so
>> freeswitc 1097       root  mem       REG        8,6       59940
>> 558034 /usr/local/freeswitch/mod/mod_say_nl.so
>> freeswitc 1097       root  mem       REG        8,6       55911
>> 558030 /usr/local/freeswitch/mod/mod_say_hu.so
>> freeswitc 1097       root  mem       REG        8,6     1675414
>> 558062 /usr/local/freeswitch/mod/mod_spidermonkey.so
>> freeswitc 1097       root  mem       REG        8,6       61373
>> 558038 /usr/local/freeswitch/mod/mod_say_th.so
>> freeswitc 1097       root  mem       REG        8,6       45694
>> 558004 /usr/local/freeswitch/mod/mod_h26x.so
>> freeswitc 1097       root  mem       REG        8,6      561707
>> 558475 /usr/local/freeswitch/lib/libnspr4.so
>> freeswitc 1097       root  mem       REG        8,6      638757
>> 557986 /usr/local/freeswitch/mod/mod_enum.so
>> freeswitc 1097       root  mem       REG        8,6      215301
>> 557984 /usr/local/freeswitch/mod/mod_dptools.so
>> freeswitc 1097       root  mem       REG        8,6     1193409
>> 558050 /usr/local/freeswitch/mod/mod_sndfile.so
>> freeswitc 1097       root  mem       REG        8,6      857708
>> 558018 /usr/local/freeswitch/mod/mod_lua.so
>> freeswitc 1097       root  mem       REG        8,6       73991
>> 558081 /usr/local/freeswitch/mod/mod_valet_parking.so
>> freeswitc 1097       root  mem       REG        8,6      196948
>> 558006 /usr/local/freeswitch/mod/mod_hash.so
>> freeswitc 1097       root  mem       REG        8,6       63432
>> 557976 /usr/local/freeswitch/mod/mod_db.so
>> freeswitc 1097       root  mem       REG        8,6     1605838
>> 558044 /usr/local/freeswitch/mod/mod_shout.so
>> freeswitc 1097       root  mem       REG        8,6       62051
>> 557962 /usr/local/freeswitch/mod/mod_cdr_csv.so
>> freeswitc 1097       root  mem       REG        8,6     5433620
>> 558052 /usr/local/freeswitch/mod/mod_sofia.so
>> freeswitc 1097       root  mem       REG        8,6      814967
>> 558058 /usr/local/freeswitch/lib/libjs.so.1.0.6
>> freeswitc 1097       root  mem       REG        8,6     2372252
>> 558054 /usr/local/freeswitch/mod/mod_spandsp.so
>> freeswitc 1097       root  mem       REG        8,6    19379720
>> 557996 /usr/local/freeswitch/mod/mod_flite.so
>> freeswitc 1097       root    3wW     REG        8,6           4
>> 525283 /usr/local/freeswitch/run/freeswitch.pid
>> freeswitc 1097       root    5u      REG        8,6      210944
>> 559634 /usr/local/freeswitch/db/core.db
>> freeswitc 1097       root    6u      REG        8,6      210944
>> 559634 /usr/local/freeswitch/db/core.db
>> freeswitc 1097       root    8u      REG        8,6     1709582
>> 559635 /usr/local/freeswitch/log/freeswitch.log
>> freeswitc 1097       root   12u      REG        8,6       97280
>> 559637 /usr/local/freeswitch/db/sofia_reg_external.db
>> freeswitc 1097       root   16u      REG        8,6       97280
>> 559638 /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
>> freeswitc 1097       root   20u      REG        8,6       97280
>> 559640 /usr/local/freeswitch/db/sofia_reg_internal.db
>> freeswitc 1097       root   37u      REG        8,6       14336
>> 559639 /usr/local/freeswitch/db/call_limit.db
>> freeswitc 1097       root   38u      REG        8,6        5120
>> 559641 /usr/local/freeswitch/db/fifo.db
>> freeswitc 1097       root   39u      REG        8,6       16384
>> 559642 /usr/local/freeswitch/db/voicemail_default.db
>> freeswitc 1097       root   41r      DIR        8,6        4096
>> 559622 /usr/local/freeswitch/sounds/music/8000
>> freeswitc 1097       root   43r      DIR        8,6        4096
>> 559622 /usr/local/freeswitch/sounds/music/8000
>> freeswitc 1097       root   47r      REG        8,6     6104850
>> 559626
>> /usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav
>> freeswitc 1097       root   48r      REG        8,6     4207128
>> 559623
>> /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav
>> fs_cli    1838       root  txt       REG        8,6      506391
>> 557912 /usr/local/freeswitch/bin/fs_cli
>> root at rajesh-desktop:~#
>>
>>
>> On Wed, Apr 18, 2012 at 1:49 AM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>
>>> wrote:
>>> > I am a newbie to freeswitch i have installed freeswitch on Ubuntu
>>> 10.04 LTS
>>> > as per installation guide, but there are no conf files in
>>> > /usr/local/freeswitch/conf folder.
>>>
>>> They might be in /opt/freeswitch/conf/
>>>
>>>
>>> > However when  i run sofia profile external
>>> > restart reloadxml in FS cli itruns successfully from where it is
>>> getting
>>> > conf files kindly help
>>>
>>> While FreeSWITCH is up, run this command: 'sudo lsof | grep
>>> freeswitch' and see what files it has open.
>>>
>>>
>>> Best,
>>> Gabe
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Wed Apr 18 15:59:35 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 18 Apr 2012 15:59:35 +0400
Subject: [Freeswitch-users] Has the behavior of loopback endpoint changed
 since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
Message-ID: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>

Today, as always, i just made git pull to the latests updates and found
strange behaviour @ vanilla configuraton of the extension
"Local_Extension".

FreeSWITCH Version 1.1.beta1 (git-e9abdac 2012-04-17 23-05-48 -0400)

 <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
        <action application="bind_meta_app" data="2 b s
record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
        <action application="bind_meta_app" data="4 b s
execute_extension::att_xfer XML features"/>
        <action application="set" data="ringback=${ru-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <!--<action application="set"
data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
-->
        <action application="set" data="continue_on_fail=true"/>
        <action application="hash"
data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set"
data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
        <action application="hash"
data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
        <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

The purpose of that extension is simple, just rings to another extension
and, for example, if the other side is busy go to voicemail. But with the
lattest git it doesn't work at all.
Why does it proceed "voicemail" into default context? Here is the log:
http://pastebin.freeswitch.com/18886

Actually it works when using 92e89cb4532913a3fd7a4888f8c94df379775cc3
revision.
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From freeswitch-list at necosec.de  Wed Apr 18 12:13:46 2012
From: freeswitch-list at necosec.de (NeCoSec - Ulrich Schinz)
Date: Wed, 18 Apr 2012 10:13:46 +0200
Subject: [Freeswitch-users] No ringback/transfer_ringback freetdm
Message-ID: <4F8E77BA.4060705@necosec.de>

Hi there,

first of all, my name is Uli. I'm reading this list for a while and
today im gonna post my first question ;)

I'm using a Sangoma A500 with FreeTDM and FreeSWITCH. My setup is
running, better say _was_ running, perfectly.
I'm not 100% sure, but maybe sunday or monday I updated freeswitch.
Since there I have a strange thing. If someone is calling via ISDN to my
FreeSWITCH-box there is no ringback tone on the callers phone (i.e.
tooot...tooot...tooot...). My phone here is ringing, I can answer the
call, after timeout voicemail answers.

Here are some parts of my config:

Extension, which is matched:

    <extension name="coming_from_isdn_bureau">
      <condition field="destination_number" expression="^(<MYNUMBER>)$">
        <action application="set"
data="effective_caller_id_number=00${caller_id_number}"/>
        <action application="set" data="ringback=$${de-ring}"/>
        <action application="set" data="transfer_ringback=$${de-ring}"/>
        <action application="transfer" data="225 XML default"/>
      </condition>
    </extension>

In my first running setup I did not set variable ringback and
transfer_ringback. And there was a toot...tooot...toot...
Observing that there is no ringback any more I decided to set these
variables, but with no (or not the wanted) effect.

Application info sais:

variable_ringback: [%(1000,4000,425)]
variable_transfer_ringback: [%(1000,4000,425)]

Calling internal (sip->sip) is no problem. Ringtones are there.
Calling from sip-phone to external number via ISDN (Sangoma A500) is
also no problem. There I don't set ringback-variable...

    <extension name="ext_isdn_calls">
      <condition field="caller_id_number" expression="^(222|223)$"
break="never">
        <action application="set"
data="effective_caller_id_number=<MSN-A>"/>
        <anti-action application="set"
data="effective_caller_id_number=<MSN-B>"/>
      </condition>
      <condition field="destination_number" expression="^(0)(\d+)$">
        <action application="info"/>
        <action application="bridge" data="freetdm/wp1/a/$2"/>
      </condition>
    </extension>

Maybe someone has an idea, where I can do further debugging or where
there could be a misconfiguration...

Best regards
Uli

P.s.: hope you can understand my german english ;)


From Stefan.Weigel at allianz-warranty.com  Wed Apr 18 17:20:20 2012
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Wed, 18 Apr 2012 15:20:20 +0200
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
	dialplans
Message-ID: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>

Hi,

 

I'm trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
content.

My static dialplan looks like:

 

 

/tmp/test.xml:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>

<document type="freeswitch/xml">

  <section name="dialplan" description="RE Dial Plan For FreeSwitch">

    <context name="public">

 

    <extension name="unloop">

      <condition field="${unroll_loops}" expression="^true$"/>

      <condition field="${sip_looped_call}" expression="^true$">

        <action application="deflect" data="${destination_number}"/>

      </condition>

    </extension>

 

    <extension name="outside_call" continue="true">

      <condition>

        <action application="set" data="outside_call=true"/>

      </condition>

    </extension>

 

    <extension name="call_debug" continue="true">

      <condition field="${call_debug}" expression="^true$" break="never">

        <action application="info"/>

      </condition>

    </extension>

 

    </context>

  </section>

</document>

 

I changed parameter "dialplan" in profile internal.xml:

 

    <param name="dialplan" value="XML:/tmp/test.xml,XML"/>

 

But when placing a call I only see that diaplan from static XML is parsed
and executed (message "has executed the last dialplan instruction, hanging
up.").

 

 

What's going wrong ?

 

 

 

Thanks and cheers

 

Stefan

 

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From mitch.capper at gmail.com  Wed Apr 18 17:56:41 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 18 Apr 2012 06:56:41 -0700
Subject: [Freeswitch-users] Has the behavior of loopback endpoint
 changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
In-Reply-To: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
References: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
Message-ID: <CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>

If you don't mind doing some more testing you can help us narrow down
exactly what commit it broke for you on and will result in the fastest
fixing of the issue.
Using git bisect will do a binary search between the known good and
known bad commits to help you find exactly what broke it in the
shortest amount of time.  To try and find the issue that you know is
broken in HEAD but fine in that revision you posted:

git bisect start HEAD 92e89cb4532913a3fd7a4888f8c94df379775cc3


then compile / test etc/
the tell git:
git bisect bad or git bisect good
then it will move on to the next
git bisect reset when done

git bisect skip if you cant test a specific revision


If you have a script that can determine if it works or not you can use:

git bisect run [my_script] [arguments]
Note that the script (my_script in the above example) should exit with
code 0 if the current source code is good, and exit with a code
between 1 and 127 (inclusive), except 125, if the current source code
is bad.

Either way file a JIRA bug report on the broken / changed functionality.

~Mitch


From bdfoster at endigotech.com  Wed Apr 18 18:25:40 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 18 Apr 2012 10:25:40 -0400
Subject: [Freeswitch-users] FreeSwitch Configuration files
In-Reply-To: <CANBhyPBS5_PYjmHmc8ns_Vmk-zAX2Q7ZYSspGKOLyniYnSnhCg@mail.gmail.com>
References: <CAAMWH7r60Y718zkNSM_HaCU=23RPdO1S++_TaP54OOWfhEhisg@mail.gmail.com>
	<CAMwZ-t+QeQpPzggHsfPOONrsRinvkJ97KCTjgFXj1+mkt3ZraQ@mail.gmail.com>
	<CAAMWH7q01R9HR9TkmD7Ga-E+VXZMAOOvn3gpgbG58Km6ZzrhfQ@mail.gmail.com>
	<CAKV-jFwQ33ujAP=o0qPmVomdPQs+aNK3EU_LXE7NA83-khwe9g@mail.gmail.com>
	<CANBhyPBS5_PYjmHmc8ns_Vmk-zAX2Q7ZYSspGKOLyniYnSnhCg@mail.gmail.com>
Message-ID: <CAKV-jFz921ARh0QSf4=tojaPdYP1+jVhXCwfba5e-LaPn_c7Kw@mail.gmail.com>

Your dialplans are here: /usr/local/freeswitch/conf/dialplan
Your directory is here: /usr/local/freeswitch/conf/directory/default
Your gateways go here: /usr/local/freeswitch/conf/sip_profiles/external
Profile settings: /usr/local/freeswitch/conf/sip_profiles/external.xml or
internal.xml or internal-ip6.xml


Please make sure you read the getting started guide on the wiki before you
ask any more questions. Thanks.

-BDF

2012/4/18 Anton Kvashenkin <anton.jugatsu at gmail.com>

> Ok. You can try
>
> strace -e trace=open /usr/local/freeswitch/bin/freeswitch 2>&1 | grep conf
>
> 18 ?????? 2012 ?. 10:38 ???????????? Brian Foster <bdfoster at endigotech.com
> > ???????:
>
> Try:
>> sudo lsof | grep default
>>
>> -BDF
>>
>>
>>
>> On Wed, Apr 18, 2012 at 12:18 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>wrote:
>>
>>> Thanks for Quick reply
>>> files are not there in  /opt/freeswitch/conf
>>> Following is the output of sudo lsof | grep freeswitch
>>>
>>> Thanks,
>>> Rajesh
>>>
>>>
>>> freeswitc 1097       root  txt       REG        8,6       37464
>>> 557911 /usr/local/freeswitch/bin/freeswitch
>>> freeswitc 1097       root  mem       REG        8,6       56818
>>> 558014 /usr/local/freeswitch/mod/mod_logfile.so
>>> freeswitc 1097       root  mem       REG        8,6       70488
>>> 558085 /usr/local/freeswitch/mod/mod_xml_cdr.so
>>> freeswitc 1097       root  mem       REG        8,6       78418
>>> 558016 /usr/local/freeswitch/mod/mod_loopback.so
>>> freeswitc 1097       root  mem       REG        8,6      123376
>>> 557992 /usr/local/freeswitch/mod/mod_expr.so
>>> freeswitc 1097       root  mem       REG        8,6       48335
>>> 557988 /usr/local/freeswitch/mod/mod_esf.so
>>> freeswitc 1097       root  mem       REG        8,6       60471
>>> 557998 /usr/local/freeswitch/mod/mod_fsv.so
>>> freeswitc 1097       root  mem       REG        8,6       65151
>>> 557980 /usr/local/freeswitch/mod/mod_dialplan_xml.so
>>> freeswitc 1097       root  mem       REG        8,6     7010899
>>> 557907 /usr/local/freeswitch/lib/libfreeswitch.so.1.0.0
>>> freeswitc 1097       root  mem       REG        8,6      234583
>>> 557968 /usr/local/freeswitch/mod/mod_commands.so
>>> freeswitc 1097       root  mem       REG        8,6      297534
>>> 557970 /usr/local/freeswitch/mod/mod_conference.so
>>> freeswitc 1097       root  mem       REG        8,6       54502
>>> 557978 /usr/local/freeswitch/mod/mod_dialplan_asterisk.so
>>> freeswitc 1097       root  mem       REG        8,6      199102
>>> 557994 /usr/local/freeswitch/mod/mod_fifo.so
>>> freeswitc 1097       root  mem       REG        8,6      255197
>>> 558083 /usr/local/freeswitch/mod/mod_voicemail.so
>>> freeswitc 1097       root  mem       REG        8,6       54099
>>> 557966 /usr/local/freeswitch/mod/mod_cluechoo.so
>>> freeswitc 1097       root  mem       REG        8,6      146786
>>> 558008 /usr/local/freeswitch/mod/mod_httapi.so
>>> freeswitc 1097       root  mem       REG        8,6       41108
>>> 558000 /usr/local/freeswitch/mod/mod_g723_1.so
>>> freeswitc 1097       root  mem       REG        8,6       40884
>>> 558002 /usr/local/freeswitch/mod/mod_g729.so
>>> freeswitc 1097       root  mem       REG        8,6       40590
>>> 557958 /usr/local/freeswitch/mod/mod_amr.so
>>> freeswitc 1097       root  mem       REG        8,6      170072
>>> 558010 /usr/local/freeswitch/mod/mod_ilbc.so
>>> freeswitc 1097       root  mem       REG        8,6       45814
>>> 558020 /usr/local/freeswitch/mod/mod_native_file.so
>>> freeswitc 1097       root  mem       REG        8,6       46476
>>> 558042 /usr/local/freeswitch/mod/mod_shell_stream.so
>>> freeswitc 1097       root  mem       REG        8,6      405014
>>> 557982 /usr/local/freeswitch/mod/mod_dingaling.so
>>> freeswitc 1097       root  mem       REG        8,6      133229
>>> 557990 /usr/local/freeswitch/mod/mod_event_socket.so
>>> freeswitc 1097       root  mem       REG        8,6      278775
>>> 558056 /usr/local/freeswitch/mod/mod_speex.so
>>> freeswitc 1097       root  mem       REG        8,6      141494
>>> 558046 /usr/local/freeswitch/mod/mod_siren.so
>>> freeswitc 1097       root  mem       REG        8,6       70598
>>> 558012 /usr/local/freeswitch/mod/mod_local_stream.so
>>> freeswitc 1097       root  mem       REG        8,6       47575
>>> 558077 /usr/local/freeswitch/mod/mod_tone_stream.so
>>> freeswitc 1097       root  mem       REG        8,6       83162
>>> 558071 /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so
>>> freeswitc 1097       root  mem       REG        8,6       58147
>>> 558069 /usr/local/freeswitch/mod/mod_spidermonkey_core_db.so
>>> freeswitc 1097       root  mem       REG        8,6       55510
>>> 558067 /usr/local/freeswitch/mod/mod_spidermonkey_socket.so
>>> freeswitc 1097       root  mem       REG        8,6       49431
>>> 558079 /usr/local/freeswitch/mod/mod_tts_commandline.so
>>> freeswitc 1097       root  mem       REG        8,6       62820
>>> 558024 /usr/local/freeswitch/mod/mod_say_en.so
>>> freeswitc 1097       root  mem       REG        8,6       60785
>>> 558040 /usr/local/freeswitch/mod/mod_say_zh.so
>>> freeswitc 1097       root  mem       REG        8,6       53090
>>> 557972 /usr/local/freeswitch/mod/mod_console.so
>>> freeswitc 1097       root  mem       REG        8,6       93667
>>> 558036 /usr/local/freeswitch/mod/mod_say_ru.so
>>> freeswitc 1097       root  mem       REG        8,6       60752
>>> 558022 /usr/local/freeswitch/mod/mod_say_de.so
>>> freeswitc 1097       root  mem       REG        8,6       60680
>>> 558026 /usr/local/freeswitch/mod/mod_say_es.so
>>> freeswitc 1097       root  mem       REG        8,6       61467
>>> 558028 /usr/local/freeswitch/mod/mod_say_fr.so
>>> freeswitc 1097       root  mem       REG        8,6       60021
>>> 558032 /usr/local/freeswitch/mod/mod_say_it.so
>>> freeswitc 1097       root  mem       REG        8,6       59940
>>> 558034 /usr/local/freeswitch/mod/mod_say_nl.so
>>> freeswitc 1097       root  mem       REG        8,6       55911
>>> 558030 /usr/local/freeswitch/mod/mod_say_hu.so
>>> freeswitc 1097       root  mem       REG        8,6     1675414
>>> 558062 /usr/local/freeswitch/mod/mod_spidermonkey.so
>>> freeswitc 1097       root  mem       REG        8,6       61373
>>> 558038 /usr/local/freeswitch/mod/mod_say_th.so
>>> freeswitc 1097       root  mem       REG        8,6       45694
>>> 558004 /usr/local/freeswitch/mod/mod_h26x.so
>>> freeswitc 1097       root  mem       REG        8,6      561707
>>> 558475 /usr/local/freeswitch/lib/libnspr4.so
>>> freeswitc 1097       root  mem       REG        8,6      638757
>>> 557986 /usr/local/freeswitch/mod/mod_enum.so
>>> freeswitc 1097       root  mem       REG        8,6      215301
>>> 557984 /usr/local/freeswitch/mod/mod_dptools.so
>>> freeswitc 1097       root  mem       REG        8,6     1193409
>>> 558050 /usr/local/freeswitch/mod/mod_sndfile.so
>>> freeswitc 1097       root  mem       REG        8,6      857708
>>> 558018 /usr/local/freeswitch/mod/mod_lua.so
>>> freeswitc 1097       root  mem       REG        8,6       73991
>>> 558081 /usr/local/freeswitch/mod/mod_valet_parking.so
>>> freeswitc 1097       root  mem       REG        8,6      196948
>>> 558006 /usr/local/freeswitch/mod/mod_hash.so
>>> freeswitc 1097       root  mem       REG        8,6       63432
>>> 557976 /usr/local/freeswitch/mod/mod_db.so
>>> freeswitc 1097       root  mem       REG        8,6     1605838
>>> 558044 /usr/local/freeswitch/mod/mod_shout.so
>>> freeswitc 1097       root  mem       REG        8,6       62051
>>> 557962 /usr/local/freeswitch/mod/mod_cdr_csv.so
>>> freeswitc 1097       root  mem       REG        8,6     5433620
>>> 558052 /usr/local/freeswitch/mod/mod_sofia.so
>>> freeswitc 1097       root  mem       REG        8,6      814967
>>> 558058 /usr/local/freeswitch/lib/libjs.so.1.0.6
>>> freeswitc 1097       root  mem       REG        8,6     2372252
>>> 558054 /usr/local/freeswitch/mod/mod_spandsp.so
>>> freeswitc 1097       root  mem       REG        8,6    19379720
>>> 557996 /usr/local/freeswitch/mod/mod_flite.so
>>> freeswitc 1097       root    3wW     REG        8,6           4
>>> 525283 /usr/local/freeswitch/run/freeswitch.pid
>>> freeswitc 1097       root    5u      REG        8,6      210944
>>> 559634 /usr/local/freeswitch/db/core.db
>>> freeswitc 1097       root    6u      REG        8,6      210944
>>> 559634 /usr/local/freeswitch/db/core.db
>>> freeswitc 1097       root    8u      REG        8,6     1709582
>>> 559635 /usr/local/freeswitch/log/freeswitch.log
>>> freeswitc 1097       root   12u      REG        8,6       97280
>>> 559637 /usr/local/freeswitch/db/sofia_reg_external.db
>>> freeswitc 1097       root   16u      REG        8,6       97280
>>> 559638 /usr/local/freeswitch/db/sofia_reg_internal-ipv6.db
>>> freeswitc 1097       root   20u      REG        8,6       97280
>>> 559640 /usr/local/freeswitch/db/sofia_reg_internal.db
>>> freeswitc 1097       root   37u      REG        8,6       14336
>>> 559639 /usr/local/freeswitch/db/call_limit.db
>>> freeswitc 1097       root   38u      REG        8,6        5120
>>> 559641 /usr/local/freeswitch/db/fifo.db
>>> freeswitc 1097       root   39u      REG        8,6       16384
>>> 559642 /usr/local/freeswitch/db/voicemail_default.db
>>> freeswitc 1097       root   41r      DIR        8,6        4096
>>> 559622 /usr/local/freeswitch/sounds/music/8000
>>> freeswitc 1097       root   43r      DIR        8,6        4096
>>> 559622 /usr/local/freeswitch/sounds/music/8000
>>> freeswitc 1097       root   47r      REG        8,6     6104850
>>> 559626
>>> /usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav
>>> freeswitc 1097       root   48r      REG        8,6     4207128
>>> 559623
>>> /usr/local/freeswitch/sounds/music/8000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav
>>> fs_cli    1838       root  txt       REG        8,6      506391
>>> 557912 /usr/local/freeswitch/bin/fs_cli
>>> root at rajesh-desktop:~#
>>>
>>>
>>> On Wed, Apr 18, 2012 at 1:49 AM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>>
>>>> On Tue, Apr 17, 2012 at 7:23 AM, Rajesh Choudhary <rajesh.mbm at gmail.com>
>>>> wrote:
>>>> > I am a newbie to freeswitch i have installed freeswitch on Ubuntu
>>>> 10.04 LTS
>>>> > as per installation guide, but there are no conf files in
>>>> > /usr/local/freeswitch/conf folder.
>>>>
>>>> They might be in /opt/freeswitch/conf/
>>>>
>>>>
>>>> > However when  i run sofia profile external
>>>> > restart reloadxml in FS cli itruns successfully from where it is
>>>> getting
>>>> > conf files kindly help
>>>>
>>>> While FreeSWITCH is up, run this command: 'sudo lsof | grep
>>>> freeswitch' and see what files it has open.
>>>>
>>>>
>>>> Best,
>>>> Gabe
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From bdfoster at endigotech.com  Wed Apr 18 18:35:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Wed, 18 Apr 2012 10:35:12 -0400
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
	dialplans
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
Message-ID: <CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>

>>   <param name="dialplan" value="XML:/tmp/test.xml,XML"/>


Not even close. Please read and re-read
http://wiki.freeswitch.org/wiki/Mod_xml_curl and specifically look at
bindings.

-BDF

On Wed, Apr 18, 2012 at 9:20 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi,****
>
> ** **
>
> I?m trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
> content.****
>
> My static dialplan looks like:****
>
> ** **
>
> ** **
>
> /tmp/test.xml:****
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>****
>
> <document type="freeswitch/xml">****
>
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">****
>
>     <context name="public">****
>
> ** **
>
>     <extension name="unloop">****
>
>       <condition field="${unroll_loops}" expression="^true$"/>****
>
>       <condition field="${sip_looped_call}" expression="^true$">****
>
>         <action application="deflect" data="${destination_number}"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
>     <extension name="outside_call" continue="true">****
>
>       <condition>****
>
>         <action application="set" data="outside_call=true"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
>     <extension name="call_debug" continue="true">****
>
>       <condition field="${call_debug}" expression="^true$" break="never">*
> ***
>
>         <action application="info"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
>     </context>****
>
>   </section>****
>
> </document>****
>
> ** **
>
> I changed parameter ?dialplan? in profile internal.xml:****
>
> ** **
>
>     <param name="dialplan" value="XML:/tmp/test.xml,XML"/>****
>
> ** **
>
> But when placing a call I only see that diaplan from static XML is parsed
> and executed (message ?has executed the last dialplan instruction, hanging
> up.?).****
>
> ** **
>
> ** **
>
> What?s going wrong ?****
>
> ** **
>
> ** **
>
> ** **
>
> Thanks and cheers****
>
> ** **
>
> Stefan****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
-------------- next part --------------
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From anthony.minessale at gmail.com  Wed Apr 18 19:22:36 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 10:22:36 -0500
Subject: [Freeswitch-users] Has the behavior of loopback endpoint
 changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
In-Reply-To: <CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>
References: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
	<CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>
Message-ID: <CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>

And again....
The mailing list is not appropriate for reporting bugs.  We have a
dedicated bug tracker http://jira.freeswitch.org
I have reproduced this problem and will fix it but I am trying to
resist the urge to refuse simply because there is no bug report I can
file the fix against.



On Wed, Apr 18, 2012 at 8:56 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
> If you don't mind doing some more testing you can help us narrow down
> exactly what commit it broke for you on and will result in the fastest
> fixing of the issue.
> Using git bisect will do a binary search between the known good and
> known bad commits to help you find exactly what broke it in the
> shortest amount of time. ?To try and find the issue that you know is
> broken in HEAD but fine in that revision you posted:
>
> git bisect start HEAD 92e89cb4532913a3fd7a4888f8c94df379775cc3
>
>
> then compile / test etc/
> the tell git:
> git bisect bad or git bisect good
> then it will move on to the next
> git bisect reset when done
>
> git bisect skip if you cant test a specific revision
>
>
> If you have a script that can determine if it works or not you can use:
>
> git bisect run [my_script] [arguments]
> Note that the script (my_script in the above example) should exit with
> code 0 if the current source code is good, and exit with a code
> between 1 and 127 (inclusive), except 125, if the current source code
> is bad.
>
> Either way file a JIRA bug report on the broken / changed functionality.
>
> ~Mitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Apr 18 19:28:53 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 10:28:53 -0500
Subject: [Freeswitch-users] Has the behavior of loopback endpoint
 changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
In-Reply-To: <CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>
References: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
	<CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>
	<CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>
Message-ID: <CAKbxfG9BnAk1BRtw=8uWR5Rnc65=VryV_oWDiCdaoAuwRsfcZA@mail.gmail.com>

commit c469321d8b8350ad6c03cd90d70c1b5b14c9b580
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Wed Apr 18 08:24:15 2012 -0500

    fix regression in commit 19f441c6f4516f035828aa112bcf92f47f4e0f28


On Wed, Apr 18, 2012 at 10:22 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> And again....
> The mailing list is not appropriate for reporting bugs. ?We have a
> dedicated bug tracker http://jira.freeswitch.org
> I have reproduced this problem and will fix it but I am trying to
> resist the urge to refuse simply because there is no bug report I can
> file the fix against.
>
>
>
> On Wed, Apr 18, 2012 at 8:56 AM, Mitch Capper <mitch.capper at gmail.com> wrote:
>> If you don't mind doing some more testing you can help us narrow down
>> exactly what commit it broke for you on and will result in the fastest
>> fixing of the issue.
>> Using git bisect will do a binary search between the known good and
>> known bad commits to help you find exactly what broke it in the
>> shortest amount of time. ?To try and find the issue that you know is
>> broken in HEAD but fine in that revision you posted:
>>
>> git bisect start HEAD 92e89cb4532913a3fd7a4888f8c94df379775cc3
>>
>>
>> then compile / test etc/
>> the tell git:
>> git bisect bad or git bisect good
>> then it will move on to the next
>> git bisect reset when done
>>
>> git bisect skip if you cant test a specific revision
>>
>>
>> If you have a script that can determine if it works or not you can use:
>>
>> git bisect run [my_script] [arguments]
>> Note that the script (my_script in the above example) should exit with
>> code 0 if the current source code is good, and exit with a code
>> between 1 and 127 (inclusive), except 125, if the current source code
>> is bad.
>>
>> Either way file a JIRA bug report on the broken / changed functionality.
>>
>> ~Mitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Apr 18 19:36:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 10:36:15 -0500
Subject: [Freeswitch-users] Call transfer failed when bypass media
In-Reply-To: <CAOTmbMyUkTjSLH2MW79j2-2E1wNkQsTYYGoR2=5WTsshKZVJuQ@mail.gmail.com>
References: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>
	<CAKbxfG9LBeBQNhtr5s-DMNt+-2s+=G-0k997Vqii2ugvhESVng@mail.gmail.com>
	<CAOTmbMyUkTjSLH2MW79j2-2E1wNkQsTYYGoR2=5WTsshKZVJuQ@mail.gmail.com>
Message-ID: <CAKbxfG8TY7OOA0SXT=8iMfe3Bth9u+X96e9h8YT3jvN7pYdNRw@mail.gmail.com>

apart from commercial support you only choices are the mailing lists and irc.
The list is very active, if you get no response you may need to ask
the question with a more descriptive subject to get the attention of
the right people.


On Wed, Apr 18, 2012 at 1:13 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> AM,
>
> Thank you very much! i'll try them.
> i ever saw these two parameters in wiki, however, i understand it the value
> of "media-option" can be either "resume-media-on-hold" or
> "bypass-media-after-att-xfer", can not configure both...so i did not try
> both yet...
>
> usually i send support request to FreeSWITCH-users at lists.freeswitch.org,
> however, most of them has no any response, could you please advise more
> efficient way, so far i know only this way for requesting support, thanks a
> lot!
>
> Regards,
> Charles
>
> 2012/4/18 Anthony Minessale <anthony.minessale at gmail.com>
>>
>> try the following 2 params in your sofia profile:
>>
>> <param name="media-option" value="resume-media-on-hold"/>
>> <param name="media-option" value="bypass-media-after-att-xfer"/>
>>
>> and use latest GIT
>>
>> Please do not direct questions to me directly, we have several
>> official methods of getting FS support and I cannot keep track of
>> random individual email threads.
>>
>>
>>
>> On Tue, Apr 17, 2012 at 8:23 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> > Dear AM,
>> >
>> > i would like to make media bypass for lower FS traffic, however, if i
>> > enable
>> > media bypass, the call transfer failed, the parameters as following,
>> >
>> > ?? <param name="inbound-bypass-media" value="true"/>
>> > ? ? <param name="inbound-proxy-media" value="false"/>
>> >
>> > but if i disable media bypass like configuration below, call transfer
>> > works
>> > well.
>> >
>> > ? ?<param name="inbound-bypass-media" value="false"/>
>> > ? ? <param name="inbound-proxy-media" value="false"/>
>> >
>> > the?scenario?is very simple, PSTN user X->media GW->FS->Registered user
>> > A, A
>> > answer the call, then press hold button and dial B, A talked with B, A
>> > hangup the call, pstn user X should talk with B.
>> >
>> > Could you please advise if anything else i missed, i below FS so strong,
>> > it
>> > must support this?scenario, i was suffered for this issue for over two
>> > years... thanks in advance.
>> >
>> > --
>> > Regards,
>> > Charles
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>
>
>
>
> --
> Regards,
> Charles
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jkomar at jbox.ca  Wed Apr 18 19:37:00 2012
From: jkomar at jbox.ca (Komar, Jason)
Date: Wed, 18 Apr 2012 09:37:00 -0600
Subject: [Freeswitch-users] Has the behavior of loopback endpoint
 changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
In-Reply-To: <CAKbxfG9BnAk1BRtw=8uWR5Rnc65=VryV_oWDiCdaoAuwRsfcZA@mail.gmail.com>
References: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
	<CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>
	<CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>
	<CAKbxfG9BnAk1BRtw=8uWR5Rnc65=VryV_oWDiCdaoAuwRsfcZA@mail.gmail.com>
Message-ID: <CA+zebiY0Vo8ttOYK5Rv8qwzX_19UuZ3057jFBdaH3AFr5R_-Qw@mail.gmail.com>

On Wed, Apr 18, 2012 at 9:28 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> commit c469321d8b8350ad6c03cd90d70c1b5b14c9b580
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: ? Wed Apr 18 08:24:15 2012 -0500
>
> ? ?fix regression in commit 19f441c6f4516f035828aa112bcf92f47f4e0f28
>

Thanks for fixing this one so quickly. I would have reported a JIRA
myself, but thought it would be more appropriate for the OP to do so.

Jason


From anthony.minessale at gmail.com  Wed Apr 18 19:38:19 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 10:38:19 -0500
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
	<CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
Message-ID: <CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>

However it's not really hurting anything since that leg is independent
and getting the CDR processed is hight priority to most people.
This is why I prefer to use mod_xml_cdr to write to disk where it's
safe then use another program to sort and deliver them to a remote
location.


On Tue, Apr 17, 2012 at 3:10 PM, Michael Collins <msc at freeswitch.org> wrote:
> Thanks for chiming in on this one. If you haven't already updated the wiki
> we'll conscript someone to do it tomorrow on our conference call.
> -MC
>
>
> On Tue, Apr 17, 2012 at 6:46 AM, Barnaby Ritchley
> <barnyritchley at hotmail.com> wrote:
>>
>> Hi Guys
>>
>> Just found that when using XML CDR module, if the script that is called
>> hangs, this will also hang the b-leg. ?We had a situation where we were
>> sending CDR data to humbuglabs and because their server was responding
>> slowly, and in some cases not at all, this was causing b legs in
>> freeswitch
>> to hang.
>>
>> we added a timeout to the curl post to humbug which has solved the issue.
>>
>> More of a note to be added in to the wiki for others for future.
>>
>> Cheers
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Apr 18 19:40:58 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 10:40:58 -0500
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
	dialplans
In-Reply-To: <CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
	<CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
Message-ID: <CAKbxfG9H54LU0b0H=4gXcbXsoc3BxPqePk=nOHxx8KOjAxVMVA@mail.gmail.com>

Your static dp probably matched something which means it will not
bother to fail over to the other one.
If even one regex is matched in your static file you'll not get to do
the other one.

If you really need it to work as-is,  add a catch-all in the end of
the static file to transfer to the XML dp.




On Wed, Apr 18, 2012 at 9:35 AM, Brian Foster <bdfoster at endigotech.com> wrote:
>
>
>>> ? <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>
>
>
> Not even close. Please read and re-read
> http://wiki.freeswitch.org/wiki/Mod_xml_curl and specifically look at
> bindings.
>
> -BDF
>
> On Wed, Apr 18, 2012 at 9:20 AM, Weigel, Stefan
> <Stefan.Weigel at allianz-warranty.com> wrote:
>>
>> Hi,
>>
>>
>>
>> I?m trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
>> content.
>>
>> My static dialplan looks like:
>>
>>
>>
>>
>>
>> /tmp/test.xml:
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>
>> <document type="freeswitch/xml">
>>
>> ? <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>>
>> ??? <context name="public">
>>
>>
>>
>> ??? <extension name="unloop">
>>
>> ????? <condition field="${unroll_loops}" expression="^true$"/>
>>
>> ????? <condition field="${sip_looped_call}" expression="^true$">
>>
>> ??????? <action application="deflect" data="${destination_number}"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> ??? <extension name="outside_call" continue="true">
>>
>> ????? <condition>
>>
>> ??????? <action application="set" data="outside_call=true"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> ??? <extension name="call_debug" continue="true">
>>
>> ????? <condition field="${call_debug}" expression="^true$" break="never">
>>
>> ??????? <action application="info"/>
>>
>> ????? </condition>
>>
>> ??? </extension>
>>
>>
>>
>> ??? </context>
>>
>> ? </section>
>>
>> </document>
>>
>>
>>
>> I changed parameter ?dialplan? in profile internal.xml:
>>
>>
>>
>> ??? <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>>
>>
>>
>> But when placing a call I only see that diaplan from static XML is parsed
>> and executed (message ?has executed the last dialplan instruction, hanging
>> up.?).
>>
>>
>>
>>
>>
>> What?s going wrong ?
>>
>>
>>
>>
>>
>>
>>
>> Thanks and cheers
>>
>>
>>
>> Stefan
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From barnyritchley at hotmail.com  Wed Apr 18 19:48:36 2012
From: barnyritchley at hotmail.com (Barnaby Ritchley)
Date: Wed, 18 Apr 2012 16:48:36 +0100
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
	<CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
	<CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>
Message-ID: <BLU0-SMTP308C20F8522F4FE898CB622BD3C0@phx.gbl>

Sure.

My reason for the note was that we were trying to debug "stuck" b-leg channels and couldnt figure out what was causing the problem.  SIP traces etc showed everything was fine, and it was a process of elimination that led us to the answer.

As this script was triggered via the XML CDR module, we were of the opinion that when the CDR was POSTed to the HTTP/S script that this was dial plan independant - IE that the dial plan would not hang waiting for the CDR script to complete, or if this did cause the dial plan to hang, we would not expect to see a channel open.

Hopefully this will help someone else who comes accross the same problem.  I am of the opinion that nothing in FS needs to be changed, just some explanation added to the Wiki.  I would do this myself, but thought it best left to someone who can use one word, where i would use 100 ;)

Brgds

On 18 Apr 2012, at 16:38, Anthony Minessale wrote:

> However it's not really hurting anything since that leg is independent
> and getting the CDR processed is hight priority to most people.
> This is why I prefer to use mod_xml_cdr to write to disk where it's
> safe then use another program to sort and deliver them to a remote
> location.
> 
> 
> On Tue, Apr 17, 2012 at 3:10 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Thanks for chiming in on this one. If you haven't already updated the wiki
>> we'll conscript someone to do it tomorrow on our conference call.
>> -MC
>> 
>> 
>> On Tue, Apr 17, 2012 at 6:46 AM, Barnaby Ritchley
>> <barnyritchley at hotmail.com> wrote:
>>> 
>>> Hi Guys
>>> 
>>> Just found that when using XML CDR module, if the script that is called
>>> hangs, this will also hang the b-leg.  We had a situation where we were
>>> sending CDR data to humbuglabs and because their server was responding
>>> slowly, and in some cases not at all, this was causing b legs in
>>> freeswitch
>>> to hang.
>>> 
>>> we added a timeout to the curl post to humbug which has solved the issue.
>>> 
>>> More of a note to be added in to the wiki for others for future.
>>> 
>>> Cheers
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From Stefan.Weigel at allianz-warranty.com  Wed Apr 18 19:11:37 2012
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Wed, 18 Apr 2012 17:11:37 +0200
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static
	(XML)	dialplans
In-Reply-To: <CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
	<CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C67BE227794@AZWSMS03.azwarranty.int>

Hi Brian,

 

I configured xml_curl.conf.xml with the bindings (using fs_curl as dynamic
backend).

 

Having this content in the static file..

 

<?xml version="1.0" encoding="UTF-8" standalone="no"?>

<document type="freeswitch/xml">

 

  <section name="dialplan" description="RE Dial Plan For FreeSwitch">

    <context name="public">

 

     <!--

      <extension name="outside_call" continue="true">

        <condition>

          <action application="set" data="outside_call=true"/>

        </condition>

      </extension>

     -->

 

    </context>

  </section>

</document>

 

..the dialplan processing continues with xml curl. But as soon as any
<action> command Is triggered, the dialplan processing doesn't execute xml
curl.

 

 

 

Best regards

 

Stefan

 

 

Von: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian
Foster
Gesendet: Mittwoch, 18. April 2012 16:35
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
dialplans

 

 

>>   <param name="dialplan" value="XML:/tmp/test.xml,XML"/>



Not even close. Please read and re-read
http://wiki.freeswitch.org/wiki/Mod_xml_curl and specifically look at
bindings.

-BDF

On Wed, Apr 18, 2012 at 9:20 AM, Weigel, Stefan
<Stefan.Weigel at allianz-warranty.com> wrote:

Hi,

 

I'm trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
content.

My static dialplan looks like:

 

 

/tmp/test.xml:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>

<document type="freeswitch/xml">

  <section name="dialplan" description="RE Dial Plan For FreeSwitch">

    <context name="public">

 

    <extension name="unloop">

      <condition field="${unroll_loops}" expression="^true$"/>

      <condition field="${sip_looped_call}" expression="^true$">

        <action application="deflect" data="${destination_number}"/>

      </condition>

    </extension>

 

    <extension name="outside_call" continue="true">

      <condition>

        <action application="set" data="outside_call=true"/>

      </condition>

    </extension>

 

    <extension name="call_debug" continue="true">

      <condition field="${call_debug}" expression="^true$" break="never">

        <action application="info"/>

      </condition>

    </extension>

 

    </context>

  </section>

</document>

 

I changed parameter "dialplan" in profile internal.xml:

 

    <param name="dialplan" value="XML:/tmp/test.xml,XML"/>

 

But when placing a call I only see that diaplan from static XML is parsed
and executed (message "has executed the last dialplan instruction, hanging
up.").

 

 

What's going wrong ?

 

 

 

Thanks and cheers

 

Stefan

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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-- 
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Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

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From freeswitch-list at necosec.de  Wed Apr 18 20:21:26 2012
From: freeswitch-list at necosec.de (NeCoSec - Ulrich Schinz)
Date: Wed, 18 Apr 2012 18:21:26 +0200
Subject: [Freeswitch-users] SOLVED Re: No ringback/transfer_ringback freetdm
In-Reply-To: <4F8E77BA.4060705@necosec.de>
References: <4F8E77BA.4060705@necosec.de>
Message-ID: <4F8EEA06.4070902@necosec.de>

Hi again,

I solved my problem. Only, I'm not sure, I understand my solution....

Like written, I'm transferring the call in context default, there is an
extension defined, where ringback was set to ${de-ring}. After
commenting this part, everything works find.

But why?? Anybody has an idea?

Kind regards
Uli

Am 18.04.2012 10:13, schrieb NeCoSec - Ulrich Schinz:
> Hi there,
>
> first of all, my name is Uli. I'm reading this list for a while and
> today im gonna post my first question ;)
>
> I'm using a Sangoma A500 with FreeTDM and FreeSWITCH. My setup is
> running, better say _was_ running, perfectly.
> I'm not 100% sure, but maybe sunday or monday I updated freeswitch.
> Since there I have a strange thing. If someone is calling via ISDN to my
> FreeSWITCH-box there is no ringback tone on the callers phone (i.e.
> tooot...tooot...tooot...). My phone here is ringing, I can answer the
> call, after timeout voicemail answers.
>
> Here are some parts of my config:
>
> Extension, which is matched:
>
>     <extension name="coming_from_isdn_bureau">
>       <condition field="destination_number" expression="^(<MYNUMBER>)$">
>         <action application="set"
> data="effective_caller_id_number=00${caller_id_number}"/>
>         <action application="set" data="ringback=$${de-ring}"/>
>         <action application="set" data="transfer_ringback=$${de-ring}"/>
>         <action application="transfer" data="225 XML default"/>
>       </condition>
>     </extension>
>
> In my first running setup I did not set variable ringback and
> transfer_ringback. And there was a toot...tooot...toot...
> Observing that there is no ringback any more I decided to set these
> variables, but with no (or not the wanted) effect.
>
> Application info sais:
>
> variable_ringback: [%(1000,4000,425)]
> variable_transfer_ringback: [%(1000,4000,425)]
>
> Calling internal (sip->sip) is no problem. Ringtones are there.
> Calling from sip-phone to external number via ISDN (Sangoma A500) is
> also no problem. There I don't set ringback-variable...
>
>     <extension name="ext_isdn_calls">
>       <condition field="caller_id_number" expression="^(222|223)$"
> break="never">
>         <action application="set"
> data="effective_caller_id_number=<MSN-A>"/>
>         <anti-action application="set"
> data="effective_caller_id_number=<MSN-B>"/>
>       </condition>
>       <condition field="destination_number" expression="^(0)(\d+)$">
>         <action application="info"/>
>         <action application="bridge" data="freetdm/wp1/a/$2"/>
>       </condition>
>     </extension>
>
> Maybe someone has an idea, where I can do further debugging or where
> there could be a misconfiguration...
>
> Best regards
> Uli
>
> P.s.: hope you can understand my german english ;)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fuji246 at gmail.com  Wed Apr 18 20:36:51 2012
From: fuji246 at gmail.com (Fu Jiantao)
Date: Thu, 19 Apr 2012 00:36:51 +0800
Subject: [Freeswitch-users] Why "Creating SIP UA for profile" all failed
Message-ID: <CAKmxLjVvG0uNx4e33ZPDgfWWxBSwRVSVZpXhecDPc8eZQmWKYw@mail.gmail.com>

Hi,

  I'm trying to run freeswitch, and seems 5060 is not listenning, I'm
running it on Windows, I'm sure that 5060 is not used by other
applications, and 130.62.240.10 is the only ipv4 address on the machine.
The external ip is different one. I've tried using 127.0.0.1 to replace the
{local_ip_v4} in vars.xml, it remains the same.

  I've seen previous solution on this, but seems there're only two
possibilities, 1) the port 5060 is already used. 2) the ip is not valid.
I'm sure they are both not in my env.

  So, what are else possibilities for such errors?

  Thanks in advance!

2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 rtp-timeout-sec [300]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-register-domain
[130.62.240.10]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-subscription-domain
[130.62.240.10]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-register-db-domain
[130.62.240.10]
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 challenge-realm [auto_from]
2012-04-19 00:33:16.796875 [NOTICE] sofia.c:4372 Started Profile internal
[sofia_reg_internal]
2012-04-19 00:33:16.796875 [DEBUG] mod_sofia.c:5193 Waiting for profiles to
start
2012-04-19 00:33:16.796875 [DEBUG] sofia.c:1747 Creating agent for internal
2012-04-19 00:33:16.828125 [ERR] sofia.c:1813 Error Creating SIP UA for
profile: external
2012-04-19 00:33:16.843750 [ERR] sofia.c:1813 Error Creating SIP UA for
profile: internal
2012-04-19 00:33:16.843750 [ERR] sofia.c:1813 Error Creating SIP UA for
profile: internal-ipv6
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From anton.jugatsu at gmail.com  Wed Apr 18 20:59:22 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 18 Apr 2012 20:59:22 +0400
Subject: [Freeswitch-users] Has the behavior of loopback endpoint
 changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
In-Reply-To: <CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>
References: <CANBhyPD8qo2CkGwnGD434czwrVeu-nsthB-edvYxU_kBkP_q5w@mail.gmail.com>
	<CALn233otp19-s5SJbMd3+MXn5rz8qHtZ5sWTxwFvXmcphtVuXA@mail.gmail.com>
	<CAKbxfG-yX5B8irfQna8Y5HQnT90zFU4gqtZxkSixhg8tOUC0Og@mail.gmail.com>
Message-ID: <CANBhyPDKJv2emJ0sdgLKaoBeM8KdvT31LD+Sjr5eQWqF12TKeA@mail.gmail.com>

Sorry guys. Done http://jira.freeswitch.org/browse/FS-4130

18 ?????? 2012 ?. 19:22 ???????????? Anthony Minessale <
anthony.minessale at gmail.com> ???????:

> And again....
> The mailing list is not appropriate for reporting bugs.  We have a
> dedicated bug tracker http://jira.freeswitch.org
> I have reproduced this problem and will fix it but I am trying to
> resist the urge to refuse simply because there is no bug report I can
> file the fix against.
>
>
>
> On Wed, Apr 18, 2012 at 8:56 AM, Mitch Capper <mitch.capper at gmail.com>
> wrote:
> > If you don't mind doing some more testing you can help us narrow down
> > exactly what commit it broke for you on and will result in the fastest
> > fixing of the issue.
> > Using git bisect will do a binary search between the known good and
> > known bad commits to help you find exactly what broke it in the
> > shortest amount of time.  To try and find the issue that you know is
> > broken in HEAD but fine in that revision you posted:
> >
> > git bisect start HEAD 92e89cb4532913a3fd7a4888f8c94df379775cc3
> >
> >
> > then compile / test etc/
> > the tell git:
> > git bisect bad or git bisect good
> > then it will move on to the next
> > git bisect reset when done
> >
> > git bisect skip if you cant test a specific revision
> >
> >
> > If you have a script that can determine if it works or not you can use:
> >
> > git bisect run [my_script] [arguments]
> > Note that the script (my_script in the above example) should exit with
> > code 0 if the current source code is good, and exit with a code
> > between 1 and 127 (inclusive), except 125, if the current source code
> > is bad.
> >
> > Either way file a JIRA bug report on the broken / changed functionality.
> >
> > ~Mitch
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.vazir at gmail.com  Wed Apr 18 21:36:11 2012
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 18 Apr 2012 22:36:11 +0500
Subject: [Freeswitch-users] creating a conference for 2 separate parked
 UUIDs, please help
In-Reply-To: <CAKzWOxVNLhNkKJCJwy=FhoTY55BCpT13bK9QLkDE1-qHGWkkBA@mail.gmail.com>
References: <CAGX+Xztq4_aCWU=Z89WTAuXiYRnK2+85anEu6sYQmRa-ceMn3Q@mail.gmail.com>
	<CAKzWOxX48ZNk7bhCnUzU22r4a=DNQKoGZOqZNXcd21W+cmwyYA@mail.gmail.com>
	<CAGX+Xzu9AfnQd2=AFsSGr8gvOBOigSF-gifd==UdztPVfmdt1A@mail.gmail.com>
	<CAKzWOxVNLhNkKJCJwy=FhoTY55BCpT13bK9QLkDE1-qHGWkkBA@mail.gmail.com>
Message-ID: <CAGX+XztuPC8b4ZC-9N5Hu+prWkSPGc9qVZz+TG7ctMTYDnNyqA@mail.gmail.com>

Cool! i saw an inline syntax for dialplan, but somehow missed that I
can use it in current case, since never actually used :)

2012/4/17 Michael Collins <msc at freeswitch.org>:
>
>
> On Tue, Apr 17, 2012 at 6:40 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>
>> Thanks Michael!
>>
>> Actually I've already managed to do it via dialplan, and I was mostly
>> thinking how to do it without dialplan at all,
>> seems in conferencing case dialplan cannot be avoided,
>
>
> Challenge accepted!
>
> uuid_transfer <uuid> conference:3000 at default inline
>
> Usual disclaimer applies - please tinker and test.
> -MC
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From acrow at integrafin.co.uk  Wed Apr 18 22:22:48 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 18 Apr 2012 19:22:48 +0100
Subject: [Freeswitch-users] Originate call into conference - no early media
Message-ID: <4F8F0678.6060906@integrafin.co.uk>

Hi,

I'm using the following dialplan snippet to call new parties into a 
conference (see : 
http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example):

<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}user/$1 &conference(${conference_name})}"/>
</condition>

<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}sofia/gateway/om.net/$1 
&conference(${conference_name})}"/>
</condition>




The page referenced implies I should get early media when I call someone 
into the conference, but when I try (either for local endpoints or those 
on a remote server via a gateway) I don't hear any ringing. I do if I 
call via a PSTN gateway.

Any ideas?

Cheers

Alex

-- 
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From shaheryarkh at googlemail.com  Wed Apr 18 22:36:12 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 18 Apr 2012 23:36:12 +0500
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
Message-ID: <CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>

Try PJSIP,

http://www.pjsip.org

Thank you.


On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Is there any recommended free/open source sip client for iPhone? Thanks.
>
> --
> Regards,
> Charles
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From fluixab at bellsouth.net  Thu Apr 19 00:25:35 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Wed, 18 Apr 2012 16:25:35 -0400
Subject: [Freeswitch-users] spx file format (speex)
Message-ID: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>

Hello,

I'd like to know if anyone managed to play speex files with extension ".spx" without converting format with sox. When I try to, I get the error message below:
2012-04-18 16:23:17.191277 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]!

If there any specific module to install (such as mod_shout for mp3 files)?

Thank you.

Bernard




From avi at avimarcus.net  Thu Apr 19 00:38:45 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 18 Apr 2012 23:38:45 +0300
Subject: [Freeswitch-users] spx file format (speex)
In-Reply-To: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>
References: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>
Message-ID: <CALNZuw=QLc3g-sGAN1+SE_v-3rteq8QJjBf0KxiwVLbb+uVSGg@mail.gmail.com>

I'm pretty sure you need .speex.

Info about how to convert, too:
http://wiki.freeswitch.org/wiki/Mod_native_file<http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding>

-Avi



On Wed, Apr 18, 2012 at 11:25 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:

> Hello,
>
> I'd like to know if anyone managed to play speex files with extension
> ".spx" without converting format with sox. When I try to, I get the error
> message below:
> 2012-04-18 16:23:17.191277 [ERR] switch_core_file.c:122 Invalid file
> format [spx] for [test.spx]!
>
> If there any specific module to install (such as mod_shout for mp3 files)?
>
> Thank you.
>
> Bernard
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gcd at i.ph  Thu Apr 19 00:46:26 2012
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 19 Apr 2012 04:46:26 +0800
Subject: [Freeswitch-users] spx file format (speex)
In-Reply-To: <CALNZuw=QLc3g-sGAN1+SE_v-3rteq8QJjBf0KxiwVLbb+uVSGg@mail.gmail.com>
References: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>
	<CALNZuw=QLc3g-sGAN1+SE_v-3rteq8QJjBf0KxiwVLbb+uVSGg@mail.gmail.com>
Message-ID: <CAHKqMb5Q0i1rCE6VD-M5Y18-rJDxCPzHfO13O8QBhV8su+_-mQ@mail.gmail.com>

.speex or .SPEEX? is it case-sensitive?

On Thu, Apr 19, 2012 at 4:38 AM, Avi Marcus <avi at avimarcus.net> wrote:

> I'm pretty sure you need .speex.
>
> Info about how to convert, too:
> http://wiki.freeswitch.org/wiki/Mod_native_file<http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding>
>
> -Avi
>
>
>
> On Wed, Apr 18, 2012 at 11:25 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:
>
>> Hello,
>>
>> I'd like to know if anyone managed to play speex files with extension
>> ".spx" without converting format with sox. When I try to, I get the error
>> message below:
>> 2012-04-18 16:23:17.191277 [ERR] switch_core_file.c:122 Invalid file
>> format [spx] for [test.spx]!
>>
>> If there any specific module to install (such as mod_shout for mp3 files)?
>>
>> Thank you.
>>
>> Bernard
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fluixab at bellsouth.net  Thu Apr 19 01:03:56 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Wed, 18 Apr 2012 17:03:56 -0400
Subject: [Freeswitch-users] spx file format (speex)
In-Reply-To: <CAHKqMb5Q0i1rCE6VD-M5Y18-rJDxCPzHfO13O8QBhV8su+_-mQ@mail.gmail.com>
References: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>
	<CALNZuw=QLc3g-sGAN1+SE_v-3rteq8QJjBf0KxiwVLbb+uVSGg@mail.gmail.com>
	<CAHKqMb5Q0i1rCE6VD-M5Y18-rJDxCPzHfO13O8QBhV8su+_-mQ@mail.gmail.com>
Message-ID: <23D54523-83D0-4C74-A2E2-B27BAB7128D1@bellsouth.net>

yes and mod_speex is loaded OK. Freeswitch does not accept the ".spx" file format extension. 



On Apr 18, 2012, at 4:46 PM, Nandy Dagondon wrote:

> .speex or .SPEEX? is it case-sensitive?
> 
> On Thu, Apr 19, 2012 at 4:38 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I'm pretty sure you need .speex.
> 
> Info about how to convert, too: http://wiki.freeswitch.org/wiki/Mod_native_file
> 
> -Avi
> 
> 
> 
> On Wed, Apr 18, 2012 at 11:25 PM, Bernard Fluixa <fluixab at bellsouth.net> wrote:
> Hello,
> 
> I'd like to know if anyone managed to play speex files with extension ".spx" without converting format with sox. When I try to, I get the error message below:
> 2012-04-18 16:23:17.191277 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]!
> 
> If there any specific module to install (such as mod_shout for mp3 files)?
> 
> Thank you.
> 
> Bernard
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lazyvirus at gmx.com  Thu Apr 19 01:29:27 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Wed, 18 Apr 2012 23:29:27 +0200
Subject: [Freeswitch-users] spx file format (speex)
In-Reply-To: <23D54523-83D0-4C74-A2E2-B27BAB7128D1@bellsouth.net>
References: <4538A832-3125-4CBB-9A96-A34CD8D05E0C@bellsouth.net>
	<CALNZuw=QLc3g-sGAN1+SE_v-3rteq8QJjBf0KxiwVLbb+uVSGg@mail.gmail.com>
	<CAHKqMb5Q0i1rCE6VD-M5Y18-rJDxCPzHfO13O8QBhV8su+_-mQ@mail.gmail.com>
	<23D54523-83D0-4C74-A2E2-B27BAB7128D1@bellsouth.net>
Message-ID: <20120418232927.4a2f927f@anubis.defcon1>

On Wed, 18 Apr 2012 17:03:56 -0400
Bernard Fluixa <fluixab at bellsouth.net> wrote:

> yes and mod_speex is loaded OK. Freeswitch does not
> accept the ".spx" file format extension. 

Just in case, in the Linux speex package you have speexdec,
which can output to sound card, .wav file or raw PCM file.

JY
-- 


From anthony.minessale at gmail.com  Thu Apr 19 06:29:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 18 Apr 2012 21:29:24 -0500
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <BLU0-SMTP308C20F8522F4FE898CB622BD3C0@phx.gbl>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
	<CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
	<CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>
	<BLU0-SMTP308C20F8522F4FE898CB622BD3C0@phx.gbl>
Message-ID: <CAKbxfG9gOv4Nm7La7xGBtGfjnpTjNWiHf_H-Dqs=JJUXT=mXxA@mail.gmail.com>

What is actually happening is the CDR state CS_REPORTING comes after
the channel is hung up.
The session is executing the post in its independent thread and it
will remain in existence until the cdr post is complete.
I think if it fails, it writes it to disk for a backup.




On Wed, Apr 18, 2012 at 10:48 AM, Barnaby Ritchley
<barnyritchley at hotmail.com> wrote:
> Sure.
>
> My reason for the note was that we were trying to debug "stuck" b-leg channels and couldnt figure out what was causing the problem. ?SIP traces etc showed everything was fine, and it was a process of elimination that led us to the answer.
>
> As this script was triggered via the XML CDR module, we were of the opinion that when the CDR was POSTed to the HTTP/S script that this was dial plan independant - IE that the dial plan would not hang waiting for the CDR script to complete, or if this did cause the dial plan to hang, we would not expect to see a channel open.
>
> Hopefully this will help someone else who comes accross the same problem. ?I am of the opinion that nothing in FS needs to be changed, just some explanation added to the Wiki. ?I would do this myself, but thought it best left to someone who can use one word, where i would use 100 ;)
>
> Brgds
>
> On 18 Apr 2012, at 16:38, Anthony Minessale wrote:
>
>> However it's not really hurting anything since that leg is independent
>> and getting the CDR processed is hight priority to most people.
>> This is why I prefer to use mod_xml_cdr to write to disk where it's
>> safe then use another program to sort and deliver them to a remote
>> location.
>>
>>
>> On Tue, Apr 17, 2012 at 3:10 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> Thanks for chiming in on this one. If you haven't already updated the wiki
>>> we'll conscript someone to do it tomorrow on our conference call.
>>> -MC
>>>
>>>
>>> On Tue, Apr 17, 2012 at 6:46 AM, Barnaby Ritchley
>>> <barnyritchley at hotmail.com> wrote:
>>>>
>>>> Hi Guys
>>>>
>>>> Just found that when using XML CDR module, if the script that is called
>>>> hangs, this will also hang the b-leg. ?We had a situation where we were
>>>> sending CDR data to humbuglabs and because their server was responding
>>>> slowly, and in some cases not at all, this was causing b legs in
>>>> freeswitch
>>>> to hang.
>>>>
>>>> we added a timeout to the curl post to humbug which has solved the issue.
>>>>
>>>> More of a note to be added in to the wiki for others for future.
>>>>
>>>> Cheers
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From fieldpeak at gmail.com  Thu Apr 19 08:22:40 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 19 Apr 2012 12:22:40 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
Message-ID: <CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>

Muhammad ,

Thank you for kindly help!
pjsip looks is a source code stack, could not be installed iPhone directly,
can you please advise one free sip client which can be installed directly
on iPhone, thank you ver much!

Regards,
Charles

2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>

> Try PJSIP,
>
> http://www.pjsip.org
>
> Thank you.
>
>
> On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Is there any recommended free/open source sip client for iPhone? Thanks.
>>
>> --
>> Regards,
>> Charles
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Charles
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From fieldpeak at gmail.com  Thu Apr 19 08:23:32 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 19 Apr 2012 12:23:32 +0800
Subject: [Freeswitch-users] Call transfer failed when bypass media
In-Reply-To: <CAKbxfG8TY7OOA0SXT=8iMfe3Bth9u+X96e9h8YT3jvN7pYdNRw@mail.gmail.com>
References: <CAOTmbMyP489D1OcNxQTCA_5mfTrvzf44VjA9cUB9xfAJOa6xgQ@mail.gmail.com>
	<CAKbxfG9LBeBQNhtr5s-DMNt+-2s+=G-0k997Vqii2ugvhESVng@mail.gmail.com>
	<CAOTmbMyUkTjSLH2MW79j2-2E1wNkQsTYYGoR2=5WTsshKZVJuQ@mail.gmail.com>
	<CAKbxfG8TY7OOA0SXT=8iMfe3Bth9u+X96e9h8YT3jvN7pYdNRw@mail.gmail.com>
Message-ID: <CAOTmbMyDguNacNtV57oV5XJ=YdKcYtPrdLuuQ_ta3TnbOx3onA@mail.gmail.com>

Thank you AM!

Regards,
Charles

2012/4/18 Anthony Minessale <anthony.minessale at gmail.com>

> apart from commercial support you only choices are the mailing lists and
> irc.
> The list is very active, if you get no response you may need to ask
> the question with a more descriptive subject to get the attention of
> the right people.
>
>
> On Wed, Apr 18, 2012 at 1:13 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> > AM,
> >
> > Thank you very much! i'll try them.
> > i ever saw these two parameters in wiki, however, i understand it the
> value
> > of "media-option" can be either "resume-media-on-hold" or
> > "bypass-media-after-att-xfer", can not configure both...so i did not try
> > both yet...
> >
> > usually i send support request to FreeSWITCH-users at lists.freeswitch.org,
> > however, most of them has no any response, could you please advise more
> > efficient way, so far i know only this way for requesting support,
> thanks a
> > lot!
> >
> > Regards,
> > Charles
> >
> > 2012/4/18 Anthony Minessale <anthony.minessale at gmail.com>
> >>
> >> try the following 2 params in your sofia profile:
> >>
> >> <param name="media-option" value="resume-media-on-hold"/>
> >> <param name="media-option" value="bypass-media-after-att-xfer"/>
> >>
> >> and use latest GIT
> >>
> >> Please do not direct questions to me directly, we have several
> >> official methods of getting FS support and I cannot keep track of
> >> random individual email threads.
> >>
> >>
> >>
> >> On Tue, Apr 17, 2012 at 8:23 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> >> > Dear AM,
> >> >
> >> > i would like to make media bypass for lower FS traffic, however, if i
> >> > enable
> >> > media bypass, the call transfer failed, the parameters as following,
> >> >
> >> >    <param name="inbound-bypass-media" value="true"/>
> >> >     <param name="inbound-proxy-media" value="false"/>
> >> >
> >> > but if i disable media bypass like configuration below, call transfer
> >> > works
> >> > well.
> >> >
> >> >    <param name="inbound-bypass-media" value="false"/>
> >> >     <param name="inbound-proxy-media" value="false"/>
> >> >
> >> > the scenario is very simple, PSTN user X->media GW->FS->Registered
> user
> >> > A, A
> >> > answer the call, then press hold button and dial B, A talked with B, A
> >> > hangup the call, pstn user X should talk with B.
> >> >
> >> > Could you please advise if anything else i missed, i below FS so
> strong,
> >> > it
> >> > must support this scenario, i was suffered for this issue for over two
> >> > years... thanks in advance.
> >> >
> >> > --
> >> > Regards,
> >> > Charles
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >
> >
> >
> >
> > --
> > Regards,
> > Charles
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
Regards,
Charles
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From mthakershi at gmail.com  Thu Apr 19 09:37:47 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Thu, 19 Apr 2012 00:37:47 -0500
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <CE89AF39-66C6-4140-AFB8-28CEBA8A6F83@visionutveckling.se>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
	<CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>
	<CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>
	<CE89AF39-66C6-4140-AFB8-28CEBA8A6F83@visionutveckling.se>
Message-ID: <CAKHK4ZMHHmq96LN9Y8NXi05gZuY_5LRxWN_unv8Mcvn2FC6NOQ@mail.gmail.com>

I checked. I get same UUID.

---------------
            //obtain new UUID from server
            string pStrNewUUID = pObjEslConn.Api("create_uuid",
null).GetBody();

            ESLevent pObjESLEvOriginate = null;
            pObjESLEvOriginate =
                pObjEslConn.Bgapi(
                "originate",
                "{" +
                "origination_uuid=" + pStrNewUUID + "," +

"ignore_early_media=true,origination_caller_id_name=..}sofia/gateway/v/1" +
aStrPhNo + " 1056 XML default",
                pStrNewUUID);

            //if this remains the same, then no need to have this logic
            string pStrJobUUID = pObjESLEvOriginate.GetHeader("Job-UUID",
-1);
---------------

At this point, pStrJobUUID and pStrNewUUID are the same value.

Is this normal?

Thanks.

Malay

On Wed, Apr 18, 2012 at 12:43 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Seems strange. These must be two different values. The first uuid is
> generated for the actual call, the second is a uuid to help you keep track
> of the background job. Are you sure they are the same, and not just very
> similar?
>
> /Peter
>
> 18 apr 2012 kl. 06:58 skrev "Malay Thakershi" <mthakershi at gmail.com
> <mailto:mthakershi at gmail.com>>:
>
> I am using exact same code as given here:
> http://wiki.freeswitch.org/wiki/Esl#Examples
>
> bgapi to invoke "originate"
>
> Thanks.
>
> Malay
>
> On Tue, Apr 17, 2012 at 6:54 PM, Avi Marcus <avi at avimarcus.net<mailto:
> avi at avimarcus.net>> wrote:
> Are you using the same bgapi or just a plain API?
> -Avi
>
>
> On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com
> <mailto:mthakershi at gmail.com>> wrote:
> Hello,
>
> Here is a code snippet from ESL example in wiki.
> http://wiki.freeswitch.org/wiki/Esl#Examples
>
> my $uuid = $con->api("create_uuid")->getBody();
> my $res =
>    $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
> my $job_uuid = $res->getHeader("Job-UUID");
>
> I use same code but in ManagedELS. But, I always get uuid and job_uuid
> same value.
>
> My question is, if it is same value then is it necessary to get job_uuid
> from res?
>
> Thanks.
>
> Malay Thakershi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4f8e476532761383613141!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4f8e476532761383613141!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From hugh at irvine.com.au  Thu Apr 19 09:47:17 2012
From: hugh at irvine.com.au (Hugh Irvine)
Date: Thu, 19 Apr 2012 15:47:17 +1000
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
	<CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
Message-ID: <735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>


Hello -

I've used both 3CX and Acrobits:

	http://www.3cx.com/products/3CXPhone-for-iPhone.html

	http://www.acrobits.cz/4/acrobits-softphone-for-iphone

3CX is free, while Acrobits is not, but Acrobits has more features.

Both work quite well.

regards

Hugh


On 19 Apr 2012, at 14:22, fieldpeak wrote:

> Muhammad ,
> 
> Thank you for kindly help!
> pjsip looks is a source code stack, could not be installed iPhone directly, can you please advise one free sip client which can be installed directly on iPhone, thank you ver much!
> 
> Regards,
> Charles
> 
> 2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>
> Try PJSIP,
> 
> http://www.pjsip.org
> 
> Thank you.
> 
> 
> On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
> Is there any recommended free/open source sip client for iPhone? Thanks.
> 
> -- 
> Regards,
> Charles
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Regards,
> Charles
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peter.olsson at visionutveckling.se  Thu Apr 19 10:09:29 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 19 Apr 2012 06:09:29 +0000
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <CAKHK4ZMHHmq96LN9Y8NXi05gZuY_5LRxWN_unv8Mcvn2FC6NOQ@mail.gmail.com>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
	<CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>
	<CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>
	<CE89AF39-66C6-4140-AFB8-28CEBA8A6F83@visionutveckling.se>,
	<CAKHK4ZMHHmq96LN9Y8NXi05gZuY_5LRxWN_unv8Mcvn2FC6NOQ@mail.gmail.com>
Message-ID: <24BB792F-3F8E-44AE-B3A5-5F0FF2B1B4F6@visionutveckling.se>

In this case yes, since you're sending the first uuid as the argument to bgapi. Try to send an empty string as the last argument for bgapi.

/Peter

19 apr 2012 kl. 07:44 skrev "Malay Thakershi" <mthakershi at gmail.com<mailto:mthakershi at gmail.com>>:

I checked. I get same UUID.

---------------
            //obtain new UUID from server
            string pStrNewUUID = pObjEslConn.Api("create_uuid", null).GetBody();

            ESLevent pObjESLEvOriginate = null;
            pObjESLEvOriginate =
                pObjEslConn.Bgapi(
                "originate",
                "{" +
                "origination_uuid=" + pStrNewUUID + "," +
                "ignore_early_media=true,origination_caller_id_name=..}sofia/gateway/v/1" + aStrPhNo + " 1056 XML default",
                pStrNewUUID);

            //if this remains the same, then no need to have this logic
            string pStrJobUUID = pObjESLEvOriginate.GetHeader("Job-UUID", -1);
---------------

At this point, pStrJobUUID and pStrNewUUID are the same value.

Is this normal?

Thanks.

Malay

On Wed, Apr 18, 2012 at 12:43 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
Seems strange. These must be two different values. The first uuid is generated for the actual call, the second is a uuid to help you keep track of the background job. Are you sure they are the same, and not just very similar?

/Peter

18 apr 2012 kl. 06:58 skrev "Malay Thakershi" <mthakershi at gmail.com<mailto:mthakershi at gmail.com><mailto:mthakershi at gmail.com<mailto:mthakershi at gmail.com>>>:

I am using exact same code as given here:  http://wiki.freeswitch.org/wiki/Esl#Examples

bgapi to invoke "originate"

Thanks.

Malay

On Tue, Apr 17, 2012 at 6:54 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net><mailto:avi at avimarcus.net<mailto:avi at avimarcus.net>>> wrote:
Are you using the same bgapi or just a plain API?
-Avi


On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com<mailto:mthakershi at gmail.com><mailto:mthakershi at gmail.com<mailto:mthakershi at gmail.com>>> wrote:
Hello,

Here is a code snippet from ESL example in wiki. http://wiki.freeswitch.org/wiki/Esl#Examples

my $uuid = $con->api("create_uuid")->getBody();
my $res =
   $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
my $job_uuid = $res->getHeader("Job-UUID");

I use same code but in ManagedELS. But, I always get uuid and job_uuid same value.

My question is, if it is same value then is it necessary to get job_uuid from res?

Thanks.

Malay Thakershi

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
http://www.freeswitchsolutions.com




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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
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Official FreeSWITCH Sites
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FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
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Official FreeSWITCH Sites
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!DSPAM:4f8e476532761383613141!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
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FreeSWITCH-users mailing list
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!DSPAM:4f8fa39e32762076332727!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
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FreeSWITCH-users mailing list
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From hkalyoncu at gmail.com  Thu Apr 19 10:21:53 2012
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Thu, 19 Apr 2012 09:21:53 +0300
Subject: [Freeswitch-users] siptrace logs
Message-ID: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>

hello,

is there any way to write siptrace output to different log file than
default log file freeswitch.log?
or is there any way to write only siptrace output (no other debug, no
switch core statements) to default log file.

i tried to write only siptrace to default log file. but when i change the
switch log level
from debug to any other level, it also disables siptrace output
automatically.

following config files have log level definitions:

console.conf.xml

    <map name="all" value="warning,err,crit,alert"/>
    <map name="switch_loadable_module_process" value="all"/>
    <map name="switch_core_state_machine.c" value="all"/>
    <map name="sofia.c"                     value="info,notice"/>

    <param name="loglevel" value="notice"/>

switch.conf.xml

    <param name="loglevel" value="debug"/>  ==> when i change this, no
siptrace output

logfile.conf.xml

   <map name="all" value="debug,info,notice,warning,err,crit,alert"/>

sofia.conf.xml

 <param name="log-level" value="0"/>
.<param name="tracelevel" value="DEBUG"/>

how should i modify these parameters to accomplish what i need?

regards,
huseyin
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From anton.jugatsu at gmail.com  Thu Apr 19 10:37:27 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 19 Apr 2012 10:37:27 +0400
Subject: [Freeswitch-users] siptrace logs
In-Reply-To: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
References: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
Message-ID: <CANBhyPA8Z98DMk9O4VXrxFRnM0CWMV1+_vzxj+1jH8R9iBoPHg@mail.gmail.com>

I think that for this purpose better to use apps like ngrep or tcpdump,
just my 2 cents.

ngrep -d eth0 -qtW byline -i t38 port 5060 and host <> -O
/tmp/t38_debug.pcap

19 ?????? 2012 ?. 10:21 ???????????? huseyin kalyoncu
<hkalyoncu at gmail.com>???????:

> hello,
>
> is there any way to write siptrace output to different log file than
> default log file freeswitch.log?
> or is there any way to write only siptrace output (no other debug, no
> switch core statements) to default log file.
>
> i tried to write only siptrace to default log file. but when i change the
> switch log level
> from debug to any other level, it also disables siptrace output
> automatically.
>
> following config files have log level definitions:
>
> console.conf.xml
>
>     <map name="all" value="warning,err,crit,alert"/>
>     <map name="switch_loadable_module_process" value="all"/>
>     <map name="switch_core_state_machine.c" value="all"/>
>     <map name="sofia.c"                     value="info,notice"/>
>
>     <param name="loglevel" value="notice"/>
>
> switch.conf.xml
>
>     <param name="loglevel" value="debug"/>  ==> when i change this, no
> siptrace output
>
> logfile.conf.xml
>
>    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
> sofia.conf.xml
>
>  <param name="log-level" value="0"/>
> .<param name="tracelevel" value="DEBUG"/>
>
> how should i modify these parameters to accomplish what i need?
>
> regards,
> huseyin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fieldpeak at gmail.com  Thu Apr 19 11:06:46 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 19 Apr 2012 15:06:46 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
	<CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
	<735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>
Message-ID: <CAOTmbMy4EyBeSrTYQArR4nbOUmZGgqZJi7RZYjo1zM1fjkuEeA@mail.gmail.com>

Hi Hugh,

Thanks for you help, 3CXphone is good to use and free one, however the
vendor  has also SIP server which competing with FS,
i'm looking for a free sip client for iPhone for my users which will  not
involve competition with FS... like CSipSImple on Anroid, it as only client
not server which is safe for my project that will not involve sip server
vendor to compete, if you know, please kindly advise, thanks a lot!

Regards,
Charles

2012/4/19 Hugh Irvine <hugh at irvine.com.au>

>
> Hello -
>
> I've used both 3CX and Acrobits:
>
>        http://www.3cx.com/products/3CXPhone-for-iPhone.html
>
>        http://www.acrobits.cz/4/acrobits-softphone-for-iphone
>
> 3CX is free, while Acrobits is not, but Acrobits has more features.
>
> Both work quite well.
>
> regards
>
> Hugh
>
>
> On 19 Apr 2012, at 14:22, fieldpeak wrote:
>
> > Muhammad ,
> >
> > Thank you for kindly help!
> > pjsip looks is a source code stack, could not be installed iPhone
> directly, can you please advise one free sip client which can be installed
> directly on iPhone, thank you ver much!
> >
> > Regards,
> > Charles
> >
> > 2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>
> > Try PJSIP,
> >
> > http://www.pjsip.org
> >
> > Thank you.
> >
> >
> > On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
> > Is there any recommended free/open source sip client for iPhone? Thanks.
> >
> > --
> > Regards,
> > Charles
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Muhammad Shahzad
> > -----------------------------------
> > CISCO Rich Media Communication Specialist (CRMCS)
> > CISCO Certified Network Associate (CCNA)
> > Cell: +92 334 422 40 88
> > MSN: shari_786pk at hotmail.com
> > Email: shaheryarkh at googlemail.com
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Regards,
> > Charles
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Charles
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From brett at launch3.net  Thu Apr 19 11:08:26 2012
From: brett at launch3.net (Brett Wilson)
Date: Thu, 19 Apr 2012 03:08:26 -0400
Subject: [Freeswitch-users] simultaneous ring not the same with ring group
	vs FIFO
Message-ID: <284e4fb52e62fb142d9036db355d844b@launch3.net>

  

Hey Guys, 

I was just curious about something. I'm using
grandstream gxp2100 phones and I noticed that when I call multiple
phones in a "ring group" (bridge to comma-separated extensions), they do
not seem to ring completely in sync. I was messing around setting up a
FIFO queue, and noticed that when multiple phones are registered into
the queue, and a call comes in, they all ring simultaneously and in
sync. 

Why is this? 
-- 

*******************************************

BRETT WILSON
IT DEPARTMENT
LAUNCH 3 VENTURES, LLC 
134 Myer Street

Hackensack, NJ 07601 
PHONE: 877.878.9134
FAX: 646.536.3866 
EMAIL:
Brett.Wilson at launch3.net [1]
AOL IM: Brett.Wilson at launch3.net
[2]
www.Launch3.net [3] 
www.Launch3telecom.com [4]

******************************************* 
 [5] [6] [7] [8] 


Links:
------
[1] mailto:Brett.Wilson at launch3.net
[2]
mailto:Brett.Wilson at launch3.net
[3] http://www.launch3.net/
[4]
http://www.launch3telecom.com/
[5]
http://launch3telecom.blogspot.com/
[6]
http://www.facebook.com/pages/Launch-3-Telecom/165610820178720
[7]
https://twitter.com/launch3telecom
[8]
https://plus.google.com/102713212642017290959
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From anton.jugatsu at gmail.com  Thu Apr 19 11:13:58 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Thu, 19 Apr 2012 11:13:58 +0400
Subject: [Freeswitch-users] simultaneous ring not the same with ring
 group vs FIFO
In-Reply-To: <284e4fb52e62fb142d9036db355d844b@launch3.net>
References: <284e4fb52e62fb142d9036db355d844b@launch3.net>
Message-ID: <CANBhyPB4YT-cMLXyOEQjd0WwqvaGiNnWcr1USLQGWi6Jgs08wA@mail.gmail.com>

It would be better to provide some debug info like dialplan snippet or
siptrace logs. To simplify this try to use fs_logger
http://wiki.freeswitch.org/wiki/Fs_logger.pl

It's awesome script, very handy.

19 ?????? 2012 ?. 11:08 ???????????? Brett Wilson <brett at launch3.net>???????:

> **
>
> Hey Guys,
>
> I was just curious about something. I'm using grandstream gxp2100 phones
> and I noticed that when I call multiple phones in a "ring group" (bridge to
> comma-separated extensions), they do not seem to ring completely in sync. I
> was messing around setting up a FIFO queue, and noticed that when multiple
> phones are registered into the queue, and a call comes in, they all ring
> simultaneously and in sync.
>
> Why is this?
> --
>
> *******************************************
> *Brett Wilson*
> *IT Department*
> *Launch 3 Ventures, LLC*
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866
> *Email:* Brett.Wilson at launch3.net
> *AOL IM:* Brett.Wilson at launch3.net
> www.Launch3.net <http://www.launch3.net/>
> www.Launch3telecom.com <http://www.launch3telecom.com/>
> *******************************************
> [image: Description: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>[image:
> Description: Description: Description: GPlus-Logo]<https://plus.google.com/102713212642017290959>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From hugh at irvine.com.au  Thu Apr 19 11:22:12 2012
From: hugh at irvine.com.au (Hugh Irvine)
Date: Thu, 19 Apr 2012 17:22:12 +1000
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMy4EyBeSrTYQArR4nbOUmZGgqZJi7RZYjo1zM1fjkuEeA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
	<CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
	<735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>
	<CAOTmbMy4EyBeSrTYQArR4nbOUmZGgqZJi7RZYjo1zM1fjkuEeA@mail.gmail.com>
Message-ID: <8DE4ACF9-68FA-4073-B257-B228694732C5@irvine.com.au>


Hello Charles -

A quick Google search for "sip phone for iphone" brings up lots of hits, as does a search of the App Store.

regards

Hugh


On 19 Apr 2012, at 17:06, fieldpeak wrote:

> Hi Hugh, 
> 
> Thanks for you help, 3CXphone is good to use and free one, however the vendor  has also SIP server which competing with FS, 
> i'm looking for a free sip client for iPhone for my users which will  not involve competition with FS... like CSipSImple on Anroid, it as only client not server which is safe for my project that will not involve sip server vendor to compete, if you know, please kindly advise, thanks a lot!
> 
> Regards,
> Charles
> 
> 2012/4/19 Hugh Irvine <hugh at irvine.com.au>
> 
> Hello -
> 
> I've used both 3CX and Acrobits:
> 
>        http://www.3cx.com/products/3CXPhone-for-iPhone.html
> 
>        http://www.acrobits.cz/4/acrobits-softphone-for-iphone
> 
> 3CX is free, while Acrobits is not, but Acrobits has more features.
> 
> Both work quite well.
> 
> regards
> 
> Hugh
> 
> 
> On 19 Apr 2012, at 14:22, fieldpeak wrote:
> 
> > Muhammad ,
> >
> > Thank you for kindly help!
> > pjsip looks is a source code stack, could not be installed iPhone directly, can you please advise one free sip client which can be installed directly on iPhone, thank you ver much!
> >
> > Regards,
> > Charles
> >
> > 2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>
> > Try PJSIP,
> >
> > http://www.pjsip.org
> >
> > Thank you.
> >
> >
> > On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
> > Is there any recommended free/open source sip client for iPhone? Thanks.
> >
> > --
> > Regards,
> > Charles
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Muhammad Shahzad
> > -----------------------------------
> > CISCO Rich Media Communication Specialist (CRMCS)
> > CISCO Certified Network Associate (CCNA)
> > Cell: +92 334 422 40 88
> > MSN: shari_786pk at hotmail.com
> > Email: shaheryarkh at googlemail.com
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Regards,
> > Charles
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> -- 
> Regards,
> Charles
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fieldpeak at gmail.com  Thu Apr 19 11:26:26 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 19 Apr 2012 15:26:26 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <8DE4ACF9-68FA-4073-B257-B228694732C5@irvine.com.au>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
	<CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
	<735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>
	<CAOTmbMy4EyBeSrTYQArR4nbOUmZGgqZJi7RZYjo1zM1fjkuEeA@mail.gmail.com>
	<8DE4ACF9-68FA-4073-B257-B228694732C5@irvine.com.au>
Message-ID: <CAOTmbMyqGTDzNfuv+NttEWe1f9tiBqLk3JEuM8JXtmgb3p2x1A@mail.gmail.com>

Hi Hugh,

Thanks for your prompt reply, it is amazing...! :)

Regards,
Charles

2012/4/19 Hugh Irvine <hugh at irvine.com.au>

>
> Hello Charles -
>
> A quick Google search for "sip phone for iphone" brings up lots of hits,
> as does a search of the App Store.
>
> regards
>
> Hugh
>
>
> On 19 Apr 2012, at 17:06, fieldpeak wrote:
>
> > Hi Hugh,
> >
> > Thanks for you help, 3CXphone is good to use and free one, however the
> vendor  has also SIP server which competing with FS,
> > i'm looking for a free sip client for iPhone for my users which will
>  not involve competition with FS... like CSipSImple on Anroid, it as only
> client not server which is safe for my project that will not involve sip
> server vendor to compete, if you know, please kindly advise, thanks a lot!
> >
> > Regards,
> > Charles
> >
> > 2012/4/19 Hugh Irvine <hugh at irvine.com.au>
> >
> > Hello -
> >
> > I've used both 3CX and Acrobits:
> >
> >        http://www.3cx.com/products/3CXPhone-for-iPhone.html
> >
> >        http://www.acrobits.cz/4/acrobits-softphone-for-iphone
> >
> > 3CX is free, while Acrobits is not, but Acrobits has more features.
> >
> > Both work quite well.
> >
> > regards
> >
> > Hugh
> >
> >
> > On 19 Apr 2012, at 14:22, fieldpeak wrote:
> >
> > > Muhammad ,
> > >
> > > Thank you for kindly help!
> > > pjsip looks is a source code stack, could not be installed iPhone
> directly, can you please advise one free sip client which can be installed
> directly on iPhone, thank you ver much!
> > >
> > > Regards,
> > > Charles
> > >
> > > 2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>
> > > Try PJSIP,
> > >
> > > http://www.pjsip.org
> > >
> > > Thank you.
> > >
> > >
> > > On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com>
> wrote:
> > > Is there any recommended free/open source sip client for iPhone?
> Thanks.
> > >
> > > --
> > > Regards,
> > > Charles
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > > --
> > > Muhammad Shahzad
> > > -----------------------------------
> > > CISCO Rich Media Communication Specialist (CRMCS)
> > > CISCO Certified Network Associate (CCNA)
> > > Cell: +92 334 422 40 88
> > > MSN: shari_786pk at hotmail.com
> > > Email: shaheryarkh at googlemail.com
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > > --
> > > Regards,
> > > Charles
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > --
> > Regards,
> > Charles
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Charles
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From djbinter at gmail.com  Thu Apr 19 11:34:22 2012
From: djbinter at gmail.com (Dorn DJBinter)
Date: Thu, 19 Apr 2012 00:34:22 -0700
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMyqGTDzNfuv+NttEWe1f9tiBqLk3JEuM8JXtmgb3p2x1A@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAFZQphziQk61PQtjTTLu7eb-rOkcVUTvYk1RxXoBU4e+e8hsWA@mail.gmail.com>
	<CAOTmbMw+=AmuSSzV4-6CVPSw36Z0=uTf-v0-_J1js2rKO__6Wg@mail.gmail.com>
	<735B678E-0560-48C6-A495-1D47F6D6C8AB@irvine.com.au>
	<CAOTmbMy4EyBeSrTYQArR4nbOUmZGgqZJi7RZYjo1zM1fjkuEeA@mail.gmail.com>
	<8DE4ACF9-68FA-4073-B257-B228694732C5@irvine.com.au>
	<CAOTmbMyqGTDzNfuv+NttEWe1f9tiBqLk3JEuM8JXtmgb3p2x1A@mail.gmail.com>
Message-ID: <274804831903498803@unknownmsgid>

I am happy with Zoiper.

-djbinter

On Apr 19, 2012, at 12:29 AM, fieldpeak <fieldpeak at gmail.com> wrote:

Hi Hugh,

Thanks for your prompt reply, it is amazing...! :)

Regards,
Charles

2012/4/19 Hugh Irvine <hugh at irvine.com.au>

>
> Hello Charles -
>
> A quick Google search for "sip phone for iphone" brings up lots of hits,
> as does a search of the App Store.
>
> regards
>
> Hugh
>
>
> On 19 Apr 2012, at 17:06, fieldpeak wrote:
>
> > Hi Hugh,
> >
> > Thanks for you help, 3CXphone is good to use and free one, however the
> vendor  has also SIP server which competing with FS,
> > i'm looking for a free sip client for iPhone for my users which will
>  not involve competition with FS... like CSipSImple on Anroid, it as only
> client not server which is safe for my project that will not involve sip
> server vendor to compete, if you know, please kindly advise, thanks a lot!
> >
> > Regards,
> > Charles
> >
> > 2012/4/19 Hugh Irvine <hugh at irvine.com.au>
> >
> > Hello -
> >
> > I've used both 3CX and Acrobits:
> >
> >        http://www.3cx.com/products/3CXPhone-for-iPhone.html
> >
> >        http://www.acrobits.cz/4/acrobits-softphone-for-iphone
> >
> > 3CX is free, while Acrobits is not, but Acrobits has more features.
> >
> > Both work quite well.
> >
> > regards
> >
> > Hugh
> >
> >
> > On 19 Apr 2012, at 14:22, fieldpeak wrote:
> >
> > > Muhammad ,
> > >
> > > Thank you for kindly help!
> > > pjsip looks is a source code stack, could not be installed iPhone
> directly, can you please advise one free sip client which can be installed
> directly on iPhone, thank you ver much!
> > >
> > > Regards,
> > > Charles
> > >
> > > 2012/4/19 Muhammad Shahzad <shaheryarkh at googlemail.com>
> > > Try PJSIP,
> > >
> > > http://www.pjsip.org
> > >
> > > Thank you.
> > >
> > >
> > > On Tue, Apr 17, 2012 at 6:27 PM, fieldpeak <fieldpeak at gmail.com>
> wrote:
> > > Is there any recommended free/open source sip client for iPhone?
> Thanks.
> > >
> > > --
> > > Regards,
> > > Charles
> > >
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > > --
> > > Muhammad Shahzad
> > > -----------------------------------
> > > CISCO Rich Media Communication Specialist (CRMCS)
> > > CISCO Certified Network Associate (CCNA)
> > > Cell: +92 334 422 40 88
> > > MSN: shari_786pk at hotmail.com
> > > Email: shaheryarkh at googlemail.com
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > > --
> > > Regards,
> > > Charles
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > --
> > Regards,
> > Charles
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Charles

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From yehavi.bourvine at gmail.com  Thu Apr 19 11:40:37 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 19 Apr 2012 10:40:37 +0300
Subject: [Freeswitch-users] how to see which extensions are defined?
In-Reply-To: <CALn233rRNRKKy8Uddf-vZ-0fs=i=QR3DYSRRPk4zc9RW9TQtfg@mail.gmail.com>
References: <CAAtWKeEu42kh7Q7_NMdsGxj3KddDBCBx0BvuP_4+FzOEim3LHw@mail.gmail.com>
	<CALNZuw=TQjLj8n-OnJmJuRyxQHOC7E+qJf90TKZkT=8hMHZGrg@mail.gmail.com>
	<CAAtWKeHhCfXsPh6uaZEs57Mv9RSj42ZPcj=ni-uq3_F43bOfRw@mail.gmail.com>
	<CANBhyPDFrz+6ZErHVtkX_KJnc3i7k6dpZXbQJyeb2GL+CiP6oA@mail.gmail.com>
	<CALn233rRNRKKy8Uddf-vZ-0fs=i=QR3DYSRRPk4zc9RW9TQtfg@mail.gmail.com>
Message-ID: <CAAtWKeGtEc+EoKCBSpHm=adD6Hu17mG8hUWL=YM_hOaG8d-Rgw@mail.gmail.com>

I've installed latest GIT from today and all is working correctly...

                  Thanks, __Yehavi:

2012/4/18 Mitch Capper <mitch.capper at gmail.com>

> To test FS parsed the user configs properly you can try and pull a
> variable from them and see if that works. There was an API command
> /dptools I believe that was added in the last few months to allow you
> to do so.
>
> ~mitch
>
> 2012/4/17 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> > root at ippbx:/usr/local/freeswitch/conf# tree directory/
> >
> > 17 ?????? 2012 ?. 22:16 ???????????? Yehavi Bourvine
> > <yehavi.bourvine at gmail.com> ???????:
> >
> >> FS does not complain, and the extension file is in the directory file...
> >>
> >>                   Thanks, __Yehavi:
> >>
> >>
> >> 2012/4/17 Avi Marcus <avi at avimarcus.net>
> >>>
> >>> If you include all your config files, then FS will complain about xml
> >>> parse errors if there are any.
> >>> So how about you try a simple grep of the config "directory" directory
> >>> for the extension -- does that help at all?
> >>>
> >>> -Avi
> >>>
> >>>
> >>> On Tue, Apr 17, 2012 at 6:22 PM, Yehavi Bourvine
> >>> <yehavi.bourvine at gmail.com> wrote:
> >>>>
> >>>> Hello,
> >>>>
> >>>>   We have the following odd thing: I've tried updating FreeSwitch
> >>>> version on my test machine two days ago and since then no extension
> can be
> >>>> registered: they all get "403 fobidden". Going back to an older
> version
> >>>> makes it work.
> >>>>
> >>>> I am missing some tool/command to see which extensions are defined
> (i.e.
> >>>> something similar to "sofia status profile XXX reg", but for defined,
> not
> >>>> only registered extensions). I would like to make sure that it reads
> >>>> correctly the extensions definitions.
> >>>>
> >>>> Any such thing?
> >>>>
> >>>> I do not open JIRA yet as I am not sure whether the problem is with
> our
> >>>> config or not.
> >>>>
> >>>>                        Thanks, __Yehavi:
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From wstephen80 at gmail.com  Thu Apr 19 11:54:35 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 19 Apr 2012 09:54:35 +0200
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
Message-ID: <CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>

Fring

On Tue, Apr 17, 2012 at 3:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Is there any recommended free/open source sip client for iPhone? Thanks.
>
> --
> Regards,
> Charles
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From me at nevian.org  Thu Apr 19 13:43:41 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 19 Apr 2012 13:43:41 +0400
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
Message-ID: <614671334828621@web9e.yandex.ru>

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From me at nevian.org  Thu Apr 19 13:47:31 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 19 Apr 2012 13:47:31 +0400
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
Message-ID: <728991334828851@web22f.yandex.ru>

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From wstephen80 at gmail.com  Thu Apr 19 14:05:39 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 19 Apr 2012 12:05:39 +0200
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <614671334828621@web9e.yandex.ru>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
	<614671334828621@web9e.yandex.ru>
Message-ID: <CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>

I'm using Fring on iPhone with my FreeSwitch. In Fring "AddOn" you can
configure a normal SIP account.

Stephen

On Thu, Apr 19, 2012 at 11:43 AM, Serge S. Yuriev <me at nevian.org> wrote:

>  I'm afraid Fring is compete complete platform which doesn't have direct
> SIP connection (all traffic goes via their servers)
>
> 19.04.2012, 11:54, "Stephen Wilde" <wstephen80 at gmail.com>:
>
> Fring
>
> On Tue, Apr 17, 2012 at 3:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
> Is there any recommended free/open source sip client for iPhone? Thanks.
>
> --
> wbr,
> Serge
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From me at nevian.org  Thu Apr 19 14:59:02 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Thu, 19 Apr 2012 14:59:02 +0400
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
	<614671334828621@web9e.yandex.ru>
	<CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>
Message-ID: <72171334833143@web10e.yandex.ru>

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From wesleyakio at tuntscorp.com  Thu Apr 19 15:33:20 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Thu, 19 Apr 2012 08:33:20 -0300
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <CAKbxfG9gOv4Nm7La7xGBtGfjnpTjNWiHf_H-Dqs=JJUXT=mXxA@mail.gmail.com>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
	<CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
	<CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>
	<BLU0-SMTP308C20F8522F4FE898CB622BD3C0@phx.gbl>
	<CAKbxfG9gOv4Nm7La7xGBtGfjnpTjNWiHf_H-Dqs=JJUXT=mXxA@mail.gmail.com>
Message-ID: <CAGTsDSvYrf3kQUGZ=eU_R=kU9vuYR30CHjKBEZY2TzdqvJYT6g@mail.gmail.com>

Out of curiosity, what happens to the physical interface if that channel is
tied to a hardware card(T1, E1)?

What if a call comes in while the channel is reporting?

On Wed, Apr 18, 2012 at 11:29 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What is actually happening is the CDR state CS_REPORTING comes after
> the channel is hung up.
> The session is executing the post in its independent thread and it
> will remain in existence until the cdr post is complete.
> I think if it fails, it writes it to disk for a backup.
>
>
>
>
> On Wed, Apr 18, 2012 at 10:48 AM, Barnaby Ritchley
> <barnyritchley at hotmail.com> wrote:
> > Sure.
> >
> > My reason for the note was that we were trying to debug "stuck" b-leg
> channels and couldnt figure out what was causing the problem.  SIP traces
> etc showed everything was fine, and it was a process of elimination that
> led us to the answer.
> >
> > As this script was triggered via the XML CDR module, we were of the
> opinion that when the CDR was POSTed to the HTTP/S script that this was
> dial plan independant - IE that the dial plan would not hang waiting for
> the CDR script to complete, or if this did cause the dial plan to hang, we
> would not expect to see a channel open.
> >
> > Hopefully this will help someone else who comes accross the same
> problem.  I am of the opinion that nothing in FS needs to be changed, just
> some explanation added to the Wiki.  I would do this myself, but thought it
> best left to someone who can use one word, where i would use 100 ;)
> >
> > Brgds
> >
> > On 18 Apr 2012, at 16:38, Anthony Minessale wrote:
> >
> >> However it's not really hurting anything since that leg is independent
> >> and getting the CDR processed is hight priority to most people.
> >> This is why I prefer to use mod_xml_cdr to write to disk where it's
> >> safe then use another program to sort and deliver them to a remote
> >> location.
> >>
> >>
> >> On Tue, Apr 17, 2012 at 3:10 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>> Thanks for chiming in on this one. If you haven't already updated the
> wiki
> >>> we'll conscript someone to do it tomorrow on our conference call.
> >>> -MC
> >>>
> >>>
> >>> On Tue, Apr 17, 2012 at 6:46 AM, Barnaby Ritchley
> >>> <barnyritchley at hotmail.com> wrote:
> >>>>
> >>>> Hi Guys
> >>>>
> >>>> Just found that when using XML CDR module, if the script that is
> called
> >>>> hangs, this will also hang the b-leg.  We had a situation where we
> were
> >>>> sending CDR data to humbuglabs and because their server was responding
> >>>> slowly, and in some cases not at all, this was causing b legs in
> >>>> freeswitch
> >>>> to hang.
> >>>>
> >>>> we added a timeout to the curl post to humbug which has solved the
> issue.
> >>>>
> >>>> More of a note to be added in to the wiki for others for future.
> >>>>
> >>>> Cheers
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From wstephen80 at gmail.com  Thu Apr 19 15:36:59 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 19 Apr 2012 13:36:59 +0200
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <72171334833143@web10e.yandex.ru>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
	<614671334828621@web9e.yandex.ru>
	<CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>
	<72171334833143@web10e.yandex.ru>
Message-ID: <CAO4LeU3++c-CZjbm0Cq=pd0aEHZ3o9H1RbvHJsjXiHXM7WUcVA@mail.gmail.com>

It require an account to their server.
The call goes probally goes via them.

On Thu, Apr 19, 2012 at 12:59 PM, Serge S. Yuriev <me at nevian.org> wrote:

> IIRC even in this case traffic goes via them, isn't it?
> At least it requires account on ther server
>
> 19.04.2012, 14:05, "Stephen Wilde" <wstephen80 at gmail.com>:
>
> I'm using Fring on iPhone with my FreeSwitch. In Fring "AddOn" you can
> configure a normal SIP account.
> Stephen
>
> On Thu, Apr 19, 2012 at 11:43 AM, Serge S. Yuriev <me at nevian.org> wrote:
>
>  I'm afraid Fring is compete complete platform which doesn't have direct
> SIP connection (all traffic goes via their servers)
>
> 19.04.2012, 11:54, "Stephen Wilde" <wstephen80 at gmail.com>:
>
> Fring
>
> On Tue, Apr 17, 2012 at 3:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
> Is there any recommended free/open source sip client for iPhone? Thanks.
>
> --
> wbr,
> Serge
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From neilp at cs.stanford.edu  Thu Apr 19 15:54:59 2012
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 19 Apr 2012 17:24:59 +0530
Subject: [Freeswitch-users] capturing alphabet characters from session:read()
Message-ID: <CAAcoRq09jXN8cHvUNOLNTZit_zMHwBmaiJPV4aaWwH_B_g=feQ@mail.gmail.com>

Hello, I have a Lua IVR app running on Sangoma PRI/FreeTDM. I am getting
some funny dtmf input from time to time where session:read() returns an
alphabet character (usually "A" when it does happen, less frequently it
returns "D") instead of a digit. I am not able to reliably reproduce it,
but posting the question here in case anyone can point me in a direction to
troubleshoot this.

Thanks in advance,
Neil
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From Daniel.Knaggs at realitysolutions.co.uk  Thu Apr 19 16:10:23 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Thu, 19 Apr 2012 12:10:23 +0000
Subject: [Freeswitch-users] capturing alphabet characters from
 session:read()
In-Reply-To: <CAAcoRq09jXN8cHvUNOLNTZit_zMHwBmaiJPV4aaWwH_B_g=feQ@mail.gmail.com>
References: <CAAcoRq09jXN8cHvUNOLNTZit_zMHwBmaiJPV4aaWwH_B_g=feQ@mail.gmail.com>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05F4E5@RSV16.realitysolutions.local>

Hi Neil,

This sounds like a problem a few of us are suffering from - look on the list for "strange DTMF tones" and you'll see my posts.

So far, I've had to raise an issue with Sangoma since they wrote FreeTDM for FreeSwitch (as suggested by Michael).  I suggest you do the same as well.





From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Patel
Sent: 19 April 2012 12:55
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] capturing alphabet characters from session:read()

Hello, I have a Lua IVR app running on Sangoma PRI/FreeTDM. I am getting some funny dtmf input from time to time where session:read() returns an alphabet character (usually "A" when it does happen, less frequently it returns "D") instead of a digit. I am not able to reliably reproduce it, but posting the question here in case anyone can point me in a direction to troubleshoot this.

Thanks in advance,
Neil


Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel:    01482 373104
Mobile: 07932 408313
Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
http://





From peter.olsson at visionutveckling.se  Thu Apr 19 16:43:23 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 19 Apr 2012 12:43:23 +0000
Subject: [Freeswitch-users] capturing alphabet characters from
 session:read()
Message-ID: <1FFF97C269757C458224B7C895F35F15094646@cantor.std.visionutv.se>

On the other hand - A and D are valid DTMF tones, so are you sure no one is sending them?

0-9, *, # and A-D are valid.

Mvh
Peter Olsson


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Daniel Knaggs
Skickat: den 19 april 2012 14:10
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] capturing alphabet characters from session:read()

Hi Neil,

This sounds like a problem a few of us are suffering from - look on the list for "strange DTMF tones" and you'll see my posts.

So far, I've had to raise an issue with Sangoma since they wrote FreeTDM for FreeSwitch (as suggested by Michael).  I suggest you do the same as well.





From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Patel
Sent: 19 April 2012 12:55
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] capturing alphabet characters from session:read()

Hello, I have a Lua IVR app running on Sangoma PRI/FreeTDM. I am getting some funny dtmf input from time to time where session:read() returns an alphabet character (usually "A" when it does happen, less frequently it returns "D") instead of a digit. I am not able to reliably reproduce it, but posting the question here in case anyone can point me in a direction to troubleshoot this.

Thanks in advance,
Neil


Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel:    01482 373104
Mobile: 07932 408313
Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
http://




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f8fff3e32761554120508!



From fieldpeak at gmail.com  Thu Apr 19 16:52:09 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 19 Apr 2012 20:52:09 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAO4LeU3++c-CZjbm0Cq=pd0aEHZ3o9H1RbvHJsjXiHXM7WUcVA@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
	<614671334828621@web9e.yandex.ru>
	<CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>
	<72171334833143@web10e.yandex.ru>
	<CAO4LeU3++c-CZjbm0Cq=pd0aEHZ3o9H1RbvHJsjXiHXM7WUcVA@mail.gmail.com>
Message-ID: <CAOTmbMySwMsyHYh3768pRXwEMJvKLP5QCfE9+jfvYUWU20iy9A@mail.gmail.com>

It is really a amazing topic, so many people involved, thank you all :)
welcome more recommendation...

Regards,
Charles

2012/4/19 Stephen Wilde <wstephen80 at gmail.com>

> It require an account to their server.
> The call goes probally goes via them.
>
> On Thu, Apr 19, 2012 at 12:59 PM, Serge S. Yuriev <me at nevian.org> wrote:
>
>> IIRC even in this case traffic goes via them, isn't it?
>> At least it requires account on ther server
>>
>> 19.04.2012, 14:05, "Stephen Wilde" <wstephen80 at gmail.com>:
>>
>> I'm using Fring on iPhone with my FreeSwitch. In Fring "AddOn" you can
>> configure a normal SIP account.
>> Stephen
>>
>> On Thu, Apr 19, 2012 at 11:43 AM, Serge S. Yuriev <me at nevian.org> wrote:
>>
>>  I'm afraid Fring is compete complete platform which doesn't have direct
>> SIP connection (all traffic goes via their servers)
>>
>> 19.04.2012, 11:54, "Stephen Wilde" <wstephen80 at gmail.com>:
>>
>> Fring
>>
>> On Tue, Apr 17, 2012 at 3:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>> Is there any recommended free/open source sip client for iPhone? Thanks.
>>
>> --
>> wbr,
>> Serge
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Charles
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From Daniel.Knaggs at realitysolutions.co.uk  Thu Apr 19 16:59:10 2012
From: Daniel.Knaggs at realitysolutions.co.uk (Daniel Knaggs)
Date: Thu, 19 Apr 2012 12:59:10 +0000
Subject: [Freeswitch-users] capturing alphabet characters from
 session:read()
In-Reply-To: <1FFF97C269757C458224B7C895F35F15094646@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15094646@cantor.std.visionutv.se>
Message-ID: <F4F8972C59EAC841B78AFF3FE28E5BBE05F51E@RSV16.realitysolutions.local>

Hi Peter,

Yes the tones are valid themselves but I'm sure my customers aren't randomly pressing A-D on the phone which doesn't even have them on it!

I'm also seeing other tones as well including the numeric values all where are falsely detected.




-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 19 April 2012 13:43
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] capturing alphabet characters from session:read()

On the other hand - A and D are valid DTMF tones, so are you sure no one is sending them?

0-9, *, # and A-D are valid.

Mvh
Peter Olsson


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Daniel Knaggs
Skickat: den 19 april 2012 14:10
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] capturing alphabet characters from session:read()

Hi Neil,

This sounds like a problem a few of us are suffering from - look on the list for "strange DTMF tones" and you'll see my posts.

So far, I've had to raise an issue with Sangoma since they wrote FreeTDM for FreeSwitch (as suggested by Michael).  I suggest you do the same as well.





From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Patel
Sent: 19 April 2012 12:55
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] capturing alphabet characters from session:read()

Hello, I have a Lua IVR app running on Sangoma PRI/FreeTDM. I am getting some funny dtmf input from time to time where session:read() returns an alphabet character (usually "A" when it does happen, less frequently it returns "D") instead of a digit. I am not able to reliably reproduce it, but posting the question here in case anyone can point me in a direction to troubleshoot this.

Thanks in advance,
Neil


Daniel Knaggs
Software Developer

Reality Solutions Ltd
1 Global Business Park
Hamburg Road
Kingston upon Hull
East Yorkshire, HU7 0AE

Tel:    01482 373104
Mobile: 07932 408313
Email:  mailto:Daniel.Knaggs at realitysolutions.co.uk
http://




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f8fff3e32761554120508!


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From william.king at quentustech.com  Thu Apr 19 10:40:20 2012
From: william.king at quentustech.com (William King)
Date: Wed, 18 Apr 2012 23:40:20 -0700
Subject: [Freeswitch-users] siptrace logs
In-Reply-To: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
References: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
Message-ID: <4F8FB354.4020304@quentustech.com>

http://wiki.freeswitch.org/wiki/Packet_Capture

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/18/2012 11:21 PM, huseyin kalyoncu wrote:
> hello,
>
> is there any way to write siptrace output to different log file than
> default log file freeswitch.log?
> or is there any way to write only siptrace output (no other debug, no
> switch core statements) to default log file.
>
> i tried to write only siptrace to default log file. but when i change
> the switch log level 
> from debug to any other level, it also disables siptrace output
> automatically. 
>
> following config files have log level definitions:
>
> console.conf.xml
>
>     <map name="all" value="warning,err,crit,alert"/>
>     <map name="switch_loadable_module_process" value="all"/>
>     <map name="switch_core_state_machine.c" value="all"/>
>     <map name="sofia.c"                     value="info,notice"/>
>
>     <param name="loglevel" value="notice"/>
>
> switch.conf.xml
>
>     <param name="loglevel" value="debug"/>  ==> when i change this, no
> siptrace output
>
> logfile.conf.xml
>
>    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
> sofia.conf.xml
>
>  <param name="log-level" value="0"/>
> .<param name="tracelevel" value="DEBUG"/>
>
> how should i modify these parameters to accomplish what i need?
>
> regards,
> huseyin
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From oh2aun at gmail.com  Thu Apr 19 11:40:44 2012
From: oh2aun at gmail.com (Mikael Nordman)
Date: Thu, 19 Apr 2012 10:40:44 +0300
Subject: [Freeswitch-users] Running an expect script from freeswitch
Message-ID: <CAKyTn5T4bcWOjFm1xzPGMjFWMeWbcmvAvrWT-CZHFX6ROarUMg@mail.gmail.com>

Hi,

I am running an expect script that is executedn from a call lua session.



The problem is that the responses come very slowly when run from within
freeswitch.

It takes maybe 5 seconds to get each response to the expect script.



I am starting the expect script from lua_mod with os.execute.

I have also tried session:execute.

The behaviour is the same in both cases.



When running the expect script within a normal shell, it executes
immediately without any delays.



What could be causing the very slow  responses to the expect script inside
freeswitch ?



Mike
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From b2m at a-cti.com  Thu Apr 19 17:35:40 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 19 Apr 2012 19:05:40 +0530
Subject: [Freeswitch-users] Running an expect script from freeswitch
In-Reply-To: <CAKyTn5T4bcWOjFm1xzPGMjFWMeWbcmvAvrWT-CZHFX6ROarUMg@mail.gmail.com>
References: <CAKyTn5T4bcWOjFm1xzPGMjFWMeWbcmvAvrWT-CZHFX6ROarUMg@mail.gmail.com>
Message-ID: <CAP2fbbcpVb=9RKbixUvo3hL68YPBaRN+qP-WvQkwJJ_ju6KR6Q@mail.gmail.com>

Did you try the below one?

api = freeswitch.API();
api:execute("system", " YOUR SYSTEM COMMAND  ");

Thanks,
Bala



On Thu, Apr 19, 2012 at 1:10 PM, Mikael Nordman <oh2aun at gmail.com> wrote:

> Hi,
>
> I am running an expect script that is executedn from a call lua session.
>
>
>
> The problem is that the responses come very slowly when run from within
> freeswitch.
>
> It takes maybe 5 seconds to get each response to the expect script.
>
>
>
> I am starting the expect script from lua_mod with os.execute.
>
> I have also tried session:execute.
>
> The behaviour is the same in both cases.
>
>
>
> When running the expect script within a normal shell, it executes
> immediately without any delays.
>
>
>
> What could be causing the very slow  responses to the expect script inside
> freeswitch ?
>
>
>
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr 19 17:39:49 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 Apr 2012 16:39:49 +0300
Subject: [Freeswitch-users] siptrace logs
In-Reply-To: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
References: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
Message-ID: <CALNZuwkRtiy8a1CNb_WMA4FXYsSNkjhE6y2xY_27=e5+_9=jzA@mail.gmail.com>

Someone else just asked me this -- how do I get the sofia siptrace included
in the log, and I couldn't find it on the wiki.
Where should I be looking..?

-Avi


On Thu, Apr 19, 2012 at 9:21 AM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:

> hello,
>
> is there any way to write siptrace output to different log file than
> default log file freeswitch.log?
> or is there any way to write only siptrace output (no other debug, no
> switch core statements) to default log file.
>
> i tried to write only siptrace to default log file. but when i change the
> switch log level
> from debug to any other level, it also disables siptrace output
> automatically.
>
> following config files have log level definitions:
>
> console.conf.xml
>
>     <map name="all" value="warning,err,crit,alert"/>
>     <map name="switch_loadable_module_process" value="all"/>
>     <map name="switch_core_state_machine.c" value="all"/>
>     <map name="sofia.c"                     value="info,notice"/>
>
>     <param name="loglevel" value="notice"/>
>
> switch.conf.xml
>
>     <param name="loglevel" value="debug"/>  ==> when i change this, no
> siptrace output
>
> logfile.conf.xml
>
>    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>
> sofia.conf.xml
>
>  <param name="log-level" value="0"/>
> .<param name="tracelevel" value="DEBUG"/>
>
> how should i modify these parameters to accomplish what i need?
>
> regards,
> huseyin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From b2m at a-cti.com  Thu Apr 19 17:49:31 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 19 Apr 2012 19:19:31 +0530
Subject: [Freeswitch-users] siptrace logs
In-Reply-To: <CALNZuwkRtiy8a1CNb_WMA4FXYsSNkjhE6y2xY_27=e5+_9=jzA@mail.gmail.com>
References: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
	<CALNZuwkRtiy8a1CNb_WMA4FXYsSNkjhE6y2xY_27=e5+_9=jzA@mail.gmail.com>
Message-ID: <CAP2fbbdcwyK4-adGH9NqX-N9UJhBRorRfF3kmC8=oLAvpfh49A@mail.gmail.com>

Hi,

Did you try the below option in your sip_profiles xml's?

 <param name="sip-trace" value="yes"/>

Thanks,
Bala



On Thu, Apr 19, 2012 at 7:09 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Someone else just asked me this -- how do I get the sofia siptrace
> included in the log, and I couldn't find it on the wiki.
> Where should I be looking..?
>
> -Avi
>
>
> On Thu, Apr 19, 2012 at 9:21 AM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:
>
>> hello,
>>
>> is there any way to write siptrace output to different log file than
>> default log file freeswitch.log?
>> or is there any way to write only siptrace output (no other debug, no
>> switch core statements) to default log file.
>>
>> i tried to write only siptrace to default log file. but when i change the
>> switch log level
>> from debug to any other level, it also disables siptrace output
>> automatically.
>>
>> following config files have log level definitions:
>>
>> console.conf.xml
>>
>>     <map name="all" value="warning,err,crit,alert"/>
>>     <map name="switch_loadable_module_process" value="all"/>
>>     <map name="switch_core_state_machine.c" value="all"/>
>>     <map name="sofia.c"                     value="info,notice"/>
>>
>>     <param name="loglevel" value="notice"/>
>>
>> switch.conf.xml
>>
>>     <param name="loglevel" value="debug"/>  ==> when i change this, no
>> siptrace output
>>
>> logfile.conf.xml
>>
>>    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>>
>> sofia.conf.xml
>>
>>  <param name="log-level" value="0"/>
>> .<param name="tracelevel" value="DEBUG"/>
>>
>> how should i modify these parameters to accomplish what i need?
>>
>> regards,
>> huseyin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avi at avimarcus.net  Thu Apr 19 18:07:43 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 Apr 2012 17:07:43 +0300
Subject: [Freeswitch-users] siptrace logs
In-Reply-To: <CAP2fbbdcwyK4-adGH9NqX-N9UJhBRorRfF3kmC8=oLAvpfh49A@mail.gmail.com>
References: <CAPf+86XONtq3SXwfLgrXJJGCnr0+k1dVu0-SONjHTM+ZZqWL3g@mail.gmail.com>
	<CALNZuwkRtiy8a1CNb_WMA4FXYsSNkjhE6y2xY_27=e5+_9=jzA@mail.gmail.com>
	<CAP2fbbdcwyK4-adGH9NqX-N9UJhBRorRfF3kmC8=oLAvpfh49A@mail.gmail.com>
Message-ID: <CALNZuwmLZb=6_qj+PAZv9y1k9qXAW3SPL6TJuRHJ0=jT3=9EZg@mail.gmail.com>

How do we toggle the logging of this manually?
"sofia global siptrace [on|off]" turns toggles it for viewing in the
console, but not logging it.

After some googling I see an option: sofia tracelevel info / sofia
tracelevel debug .. but I thought I had used something different last time.

-Avi


On Thu, Apr 19, 2012 at 4:49 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Hi,
>
> Did you try the below option in your sip_profiles xml's?
>
>  <param name="sip-trace" value="yes"/>
>
> Thanks,
> Bala
>
>
>
> On Thu, Apr 19, 2012 at 7:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Someone else just asked me this -- how do I get the sofia siptrace
>> included in the log, and I couldn't find it on the wiki.
>> Where should I be looking..?
>>
>> -Avi
>>
>>
>> On Thu, Apr 19, 2012 at 9:21 AM, huseyin kalyoncu <hkalyoncu at gmail.com>wrote:
>>
>>> hello,
>>>
>>> is there any way to write siptrace output to different log file than
>>> default log file freeswitch.log?
>>> or is there any way to write only siptrace output (no other debug, no
>>> switch core statements) to default log file.
>>>
>>> i tried to write only siptrace to default log file. but when i change
>>> the switch log level
>>> from debug to any other level, it also disables siptrace output
>>> automatically.
>>>
>>> following config files have log level definitions:
>>>
>>> console.conf.xml
>>>
>>>     <map name="all" value="warning,err,crit,alert"/>
>>>     <map name="switch_loadable_module_process" value="all"/>
>>>     <map name="switch_core_state_machine.c" value="all"/>
>>>     <map name="sofia.c"                     value="info,notice"/>
>>>
>>>     <param name="loglevel" value="notice"/>
>>>
>>> switch.conf.xml
>>>
>>>     <param name="loglevel" value="debug"/>  ==> when i change this, no
>>> siptrace output
>>>
>>> logfile.conf.xml
>>>
>>>    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
>>>
>>> sofia.conf.xml
>>>
>>>  <param name="log-level" value="0"/>
>>> .<param name="tracelevel" value="DEBUG"/>
>>>
>>> how should i modify these parameters to accomplish what i need?
>>>
>>> regards,
>>> huseyin
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From b2m at a-cti.com  Thu Apr 19 19:08:49 2012
From: b2m at a-cti.com (Balamurugan Mahendran)
Date: Thu, 19 Apr 2012 20:38:49 +0530
Subject: [Freeswitch-users] Framing Error Reading!
Message-ID: <CAP2fbbeiZ5z5TPd_7qJKFkj+i2B8vQQj7gymZ+_ByZ5M44_XSw@mail.gmail.com>

Hi all,

Can some one help me to resolve 0kb recording?


2012-04-19 15:04:17.718894 [ERR] switch_core_media_bug.c:235 Framing Error
Reading!

Thanks,
Bala
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From peter.olsson at visionutveckling.se  Thu Apr 19 19:29:57 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 19 Apr 2012 15:29:57 +0000
Subject: [Freeswitch-users] Framing Error Reading!
In-Reply-To: <CAP2fbbeiZ5z5TPd_7qJKFkj+i2B8vQQj7gymZ+_ByZ5M44_XSw@mail.gmail.com>
References: <CAP2fbbeiZ5z5TPd_7qJKFkj+i2B8vQQj7gymZ+_ByZ5M44_XSw@mail.gmail.com>
Message-ID: <136AE35C-27C7-48E9-9140-A54EE26E4D3F@visionutveckling.se>

Is this on git head? If not, update to latest, or else pastebin logs for the entire call, and we'll go from there.

/Peter

19 apr 2012 kl. 17:15 skrev "Balamurugan Mahendran" <b2m at a-cti.com>:

> Hi all,
> 
> Can some one help me to resolve 0kb recording?
> 
> 
> 2012-04-19 15:04:17.718894 [ERR] switch_core_media_bug.c:235 Framing Error Reading!
> 
> Thanks,
> Bala
> 
> !DSPAM:4f90298f32766477921471!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> !DSPAM:4f90298f32766477921471!


From fluixab at bellsouth.net  Thu Apr 19 19:38:31 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Thu, 19 Apr 2012 11:38:31 -0400
Subject: [Freeswitch-users] Speex file format
Message-ID: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>

Hello,

I cannot play ".spx" (speex) file in commands such as originate? &playback(file.spx). I double-checked that mod_speex is loaded OK. Freeswitch returns

2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]!

I converted file to wav format and could play it (speexdec test.spx test.wav) with playback command. I probably missed a point here but I would like to know how I can play speex files without having to convert them to wav first. 

Anyone?

Thank you

Bernard  

From oh2aun at gmail.com  Thu Apr 19 17:57:33 2012
From: oh2aun at gmail.com (Mikael Nordman)
Date: Thu, 19 Apr 2012 16:57:33 +0300
Subject: [Freeswitch-users] Running an expect script from freeswitch
In-Reply-To: <CAP2fbbcpVb=9RKbixUvo3hL68YPBaRN+qP-WvQkwJJ_ju6KR6Q@mail.gmail.com>
References: <CAKyTn5T4bcWOjFm1xzPGMjFWMeWbcmvAvrWT-CZHFX6ROarUMg@mail.gmail.com>
	<CAP2fbbcpVb=9RKbixUvo3hL68YPBaRN+qP-WvQkwJJ_ju6KR6Q@mail.gmail.com>
Message-ID: <CAKyTn5QNj_YpA=f2Uqhp7ASrLuVtCOfZa4kScROSOT75OHs5eA@mail.gmail.com>

Thanks Bala.
It works.

Can you explain why instantiating a new API makes expect and telnet connect
to each
other in the right manner ?

Thanks again
/Mike

On Thu, Apr 19, 2012 at 4:35 PM, Balamurugan Mahendran <b2m at a-cti.com>wrote:

> Did you try the below one?
>
> api = freeswitch.API();
> api:execute("system", " YOUR SYSTEM COMMAND  ");
>
> Thanks,
> Bala
>
>
>
> On Thu, Apr 19, 2012 at 1:10 PM, Mikael Nordman <oh2aun at gmail.com> wrote:
>
>> Hi,
>>
>> I am running an expect script that is executedn from a call lua session.
>>
>>
>>
>> The problem is that the responses come very slowly when run from within
>> freeswitch.
>>
>> It takes maybe 5 seconds to get each response to the expect script.
>>
>>
>>
>> I am starting the expect script from lua_mod with os.execute.
>>
>> I have also tried session:execute.
>>
>> The behaviour is the same in both cases.
>>
>>
>>
>> When running the expect script within a normal shell, it executes
>> immediately without any delays.
>>
>>
>>
>> What could be causing the very slow  responses to the expect script
>> inside freeswitch ?
>>
>>
>>
>> Mike
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Apr 19 19:47:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 Apr 2012 08:47:44 -0700
Subject: [Freeswitch-users] capturing alphabet characters from
	session:read()
In-Reply-To: <F4F8972C59EAC841B78AFF3FE28E5BBE05F51E@RSV16.realitysolutions.local>
References: <1FFF97C269757C458224B7C895F35F15094646@cantor.std.visionutv.se>
	<F4F8972C59EAC841B78AFF3FE28E5BBE05F51E@RSV16.realitysolutions.local>
Message-ID: <CAKzWOxXMTqTD32uxn=FZkAd5ocB4R+_s+O8AaMG0Er1uFY8G2Q@mail.gmail.com>

On Thu, Apr 19, 2012 at 5:59 AM, Daniel Knaggs <
Daniel.Knaggs at realitysolutions.co.uk> wrote:

> Hi Peter,
>
> Yes the tones are valid themselves but I'm sure my customers aren't
> randomly pressing A-D on the phone which doesn't even have them on it!
>
> I'm also seeing other tones as well including the numeric values all where
> are falsely detected.
>
> It is 100% safe to assume that Joe Q Consumer is not dialing DTMF A, B, C,
or D. This is definitely an issue somewhere in FreeTDM.

-MC
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From avi at avimarcus.net  Thu Apr 19 19:50:06 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 Apr 2012 18:50:06 +0300
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
Message-ID: <CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>

Didn't we answer this yesterday? Bernard, have you tried it yet with a
.SPEEX (or .speex?) extension?

-Avi



On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:

> Hello,
>
> I cannot play ".spx" (speex) file in commands such as originate?
> &playback(file.spx). I double-checked that mod_speex is loaded OK.
> Freeswitch returns
>
> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
> format [spx] for [test.spx]!
>
> I converted file to wav format and could play it (speexdec test.spx
> test.wav) with playback command. I probably missed a point here but I would
> like to know how I can play speex files without having to convert them to
> wav first.
>
> Anyone?
>
> Thank you
>
> Bernard
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Thu Apr 19 19:50:39 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 Apr 2012 08:50:39 -0700
Subject: [Freeswitch-users] XML CDR Causes B Leg Hang
In-Reply-To: <CAGTsDSvYrf3kQUGZ=eU_R=kU9vuYR30CHjKBEZY2TzdqvJYT6g@mail.gmail.com>
References: <mailman.7420.1334669765.13443.freeswitch-users@lists.freeswitch.org>
	<BLU0-SMTP155F70671250902ACF29D9FBD3F0@phx.gbl>
	<CAKzWOxWNnu3zJsmc5ZNRv69rfEC--QMjg77QxTDrYLH_gsk+Ew@mail.gmail.com>
	<CAKbxfG9tZcac_fasqkH1yfepYFnbOwxPLTWD7c0F2CNa48uT7w@mail.gmail.com>
	<BLU0-SMTP308C20F8522F4FE898CB622BD3C0@phx.gbl>
	<CAKbxfG9gOv4Nm7La7xGBtGfjnpTjNWiHf_H-Dqs=JJUXT=mXxA@mail.gmail.com>
	<CAGTsDSvYrf3kQUGZ=eU_R=kU9vuYR30CHjKBEZY2TzdqvJYT6g@mail.gmail.com>
Message-ID: <CAKzWOxVQimLrQ9sMMX1g1zHPtXyQV=zvoheRWqSYYAXe7ndG7g@mail.gmail.com>

On Thu, Apr 19, 2012 at 4:33 AM, Wesley Akio <wesleyakio at tuntscorp.com>wrote:

> Out of curiosity, what happens to the physical interface if that channel
> is tied to a hardware card(T1, E1)?
>
> What if a call comes in while the channel is reporting?
>

CS_REPORTING means that the channel has been disconnected already, so there
is no physical port or channel being locked up. You can verify this for
yourself by watching all the CHANNEL_xxx events for a given call.

-MC
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From fluixab at bellsouth.net  Thu Apr 19 20:25:28 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Thu, 19 Apr 2012 12:25:28 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
Message-ID: <16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>

Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM Freeswitch and could play it with playback command. No problem. However, it is my understanding that speex format files have ".spx" extension. Why doesn't Freeswitch allows for ".spx" extension?  

.speex extension are not accepted. 
 
Thanks again

Bernard
  
On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:

> Didn't we answer this yesterday? Bernard, have you tried it yet with a .SPEEX (or .speex?) extension?
> 
> -Avi
> 
> 
> 
> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net> wrote:
> Hello,
> 
> I cannot play ".spx" (speex) file in commands such as originate? &playback(file.spx). I double-checked that mod_speex is loaded OK. Freeswitch returns
> 
> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]!
> 
> I converted file to wav format and could play it (speexdec test.spx test.wav) with playback command. I probably missed a point here but I would like to know how I can play speex files without having to convert them to wav first.
> 
> Anyone?
> 
> Thank you
> 
> Bernard
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu Apr 19 20:45:14 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 11:45:14 -0500
Subject: [Freeswitch-users] ESL code question
In-Reply-To: <24BB792F-3F8E-44AE-B3A5-5F0FF2B1B4F6@visionutveckling.se>
References: <CAKHK4ZMu5YW+=3otE37gAacNFzetUx4Mw-+1Y5Riuu+YVdw=BQ@mail.gmail.com>
	<CALNZuwkWDckL3nhUXA8qytjFwjUpvOw1rvKEs+0auUQWqh_x5A@mail.gmail.com>
	<CAKHK4ZOriSSx+TmK1YJoYn+aFx1pvHbFnJDrPvCXZ1YAUR1HnQ@mail.gmail.com>
	<CE89AF39-66C6-4140-AFB8-28CEBA8A6F83@visionutveckling.se>
	<CAKHK4ZMHHmq96LN9Y8NXi05gZuY_5LRxWN_unv8Mcvn2FC6NOQ@mail.gmail.com>
	<24BB792F-3F8E-44AE-B3A5-5F0FF2B1B4F6@visionutveckling.se>
Message-ID: <CAKbxfG9hRFC--oDUjSKLzQRVMQFFGsMGZZDPUf5nCj12HqxXOQ@mail.gmail.com>

when you supply pStrNewUUID as the 3rd arg to bgapi() you are
submitting it to be a manually specified job-uuid if you left it blank
a random one would be chosen.

ESLevent *ESLconnection::bgapi(const char *cmd, const char *arg, const
char *job_uuid)





On Thu, Apr 19, 2012 at 1:09 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> In this case yes, since you're sending the first uuid as the argument to bgapi. Try to send an empty string as the last argument for bgapi.
>
> /Peter
>
> 19 apr 2012 kl. 07:44 skrev "Malay Thakershi" <mthakershi at gmail.com<mailto:mthakershi at gmail.com>>:
>
> I checked. I get same UUID.
>
> ---------------
> ? ? ? ? ? ?//obtain new UUID from server
> ? ? ? ? ? ?string pStrNewUUID = pObjEslConn.Api("create_uuid", null).GetBody();
>
> ? ? ? ? ? ?ESLevent pObjESLEvOriginate = null;
> ? ? ? ? ? ?pObjESLEvOriginate =
> ? ? ? ? ? ? ? ?pObjEslConn.Bgapi(
> ? ? ? ? ? ? ? ?"originate",
> ? ? ? ? ? ? ? ?"{" +
> ? ? ? ? ? ? ? ?"origination_uuid=" + pStrNewUUID + "," +
> ? ? ? ? ? ? ? ?"ignore_early_media=true,origination_caller_id_name=..}sofia/gateway/v/1" + aStrPhNo + " 1056 XML default",
> ? ? ? ? ? ? ? ?pStrNewUUID);
>
> ? ? ? ? ? ?//if this remains the same, then no need to have this logic
> ? ? ? ? ? ?string pStrJobUUID = pObjESLEvOriginate.GetHeader("Job-UUID", -1);
> ---------------
>
> At this point, pStrJobUUID and pStrNewUUID are the same value.
>
> Is this normal?
>
> Thanks.
>
> Malay
>
> On Wed, Apr 18, 2012 at 12:43 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
> Seems strange. These must be two different values. The first uuid is generated for the actual call, the second is a uuid to help you keep track of the background job. Are you sure they are the same, and not just very similar?
>
> /Peter
>
> 18 apr 2012 kl. 06:58 skrev "Malay Thakershi" <mthakershi at gmail.com<mailto:mthakershi at gmail.com><mailto:mthakershi at gmail.com<mailto:mthakershi at gmail.com>>>:
>
> I am using exact same code as given here: ?http://wiki.freeswitch.org/wiki/Esl#Examples
>
> bgapi to invoke "originate"
>
> Thanks.
>
> Malay
>
> On Tue, Apr 17, 2012 at 6:54 PM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net><mailto:avi at avimarcus.net<mailto:avi at avimarcus.net>>> wrote:
> Are you using the same bgapi or just a plain API?
> -Avi
>
>
> On Wed, Apr 18, 2012 at 2:30 AM, Malay Thakershi <mthakershi at gmail.com<mailto:mthakershi at gmail.com><mailto:mthakershi at gmail.com<mailto:mthakershi at gmail.com>>> wrote:
> Hello,
>
> Here is a code snippet from ESL example in wiki. http://wiki.freeswitch.org/wiki/Esl#Examples
>
> my $uuid = $con->api("create_uuid")->getBody();
> my $res =
> ? $con->bgapi("originate","{origination_uuid=$uuid}$target 9664");
> my $job_uuid = $res->getHeader("Job-UUID");
>
> I use same code but in ManagedELS. But, I always get uuid and job_uuid same value.
>
> My question is, if it is same value then is it necessary to get job_uuid from res?
>
> Thanks.
>
> Malay Thakershi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
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> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org<mailto:consulting at freeswitch.org><mailto:consulting at freeswitch.org<mailto:consulting at freeswitch.org>>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org><mailto:FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
> !DSPAM:4f8e476532761383613141!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> !DSPAM:4f8fa39e32762076332727!
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4f8fa39e32762076332727!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Thu Apr 19 20:59:09 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 Apr 2012 19:59:09 +0300
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
Message-ID: <CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>

It seems all the codecs are are chosen in uppercase, as it seems to be
listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
You can probably modify mod_speex to add an alias to load .spx files also.

-Avi


On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:

> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
> Freeswitch and could play it with playback command. No problem. However, it
> is my understanding that speex format files have ".spx" extension. Why
> doesn't Freeswitch allows for ".spx" extension?
>
> .speex extension are not accepted.
>
> Thanks again
>
> Bernard
>
> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>
> Didn't we answer this yesterday? Bernard, have you tried it yet with a
> .SPEEX (or .speex?) extension?
>
> -Avi
>
>
>
> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:
>
>> Hello,
>>
>> I cannot play ".spx" (speex) file in commands such as originate?
>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>> Freeswitch returns
>>
>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>> format [spx] for [test.spx]!
>>
>> I converted file to wav format and could play it (speexdec test.spx
>> test.wav) with playback command. I probably missed a point here but I would
>> like to know how I can play speex files without having to convert them to
>> wav first.
>>
>> Anyone?
>>
>> Thank you
>>
>> Bernard
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gabe at gundy.org  Thu Apr 19 21:18:58 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 19 Apr 2012 11:18:58 -0600
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <CB5E9EFA.2D961%krice@freeswitch.org>
References: <AczqcbB76+HUOZ/87U6V5nsfrwA3nQ==>
	<CB5E9EFA.2D961%krice@freeswitch.org>
Message-ID: <CAMwZ-t+4TEAxTrkAMa3=fKuTBa23fzCvSRzMQPG1AH4fzwLx-g@mail.gmail.com>

On Mon, Feb 13, 2012 at 10:05 AM, Ken Rice <krice at freeswitch.org> wrote:
> Check it out... Yum is on the way for new freeswitch goodness.

I have to admit that I'm loving this. It makes getting up and going so
much better. Thanks for all the great work in this area!


Best,
Gabe


From anthony.minessale at gmail.com  Thu Apr 19 21:19:49 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 12:19:49 -0500
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
Message-ID: <CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>

no you, cant modify it.  This is a function of mod_native_file and its
limited to matching the iananames they registered the codec with.



On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
> It seems all the codecs are are chosen in uppercase, as it seems to be
> listed here:?http://wiki.freeswitch.org/wiki/Mod_native_file
> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
> You can probably modify mod_speex to add an alias to load .spx files also.
>
> -Avi
>
>
> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
> wrote:
>>
>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
>> Freeswitch and could play it with playback command. No problem. However, it
>> is my understanding that speex format files have ".spx" extension. Why
>> doesn't Freeswitch allows for ".spx" extension?
>>
>> .speex extension are not accepted.
>>
>> Thanks again
>>
>> Bernard
>>
>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>>
>> Didn't we answer this yesterday??Bernard, have you tried it yet with a
>> .SPEEX (or .speex?) extension?
>>
>> -Avi
>>
>>
>>
>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> wrote:
>>>
>>> Hello,
>>>
>>> I cannot play ".spx" (speex) file in commands such as originate?
>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>>> Freeswitch returns
>>>
>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>>> format [spx] for [test.spx]!
>>>
>>> I converted file to wav format and could play it (speexdec test.spx
>>> test.wav) with playback command. I probably missed a point here but I would
>>> like to know how I can play speex files without having to convert them to
>>> wav first.
>>>
>>> Anyone?
>>>
>>> Thank you
>>>
>>> Bernard
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From fluixab at bellsouth.net  Thu Apr 19 21:29:54 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Thu, 19 Apr 2012 13:29:54 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
Message-ID: <BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>

There's still something I'm not clear with. Here's what I did

1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
2) Call a user with x-Lite soft phone: originate user/1001 &playback(/home/sounds/a.SPEEX)

==> file is open and played but one can hear only noise squelch. What am I doing wrong?

FYI, I then converted a.SPEEX file back to wav format and could play that wav file.






On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:

> no you, cant modify it.  This is a function of mod_native_file and its
> limited to matching the iananames they registered the codec with.
> 
> 
> 
> On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> It seems all the codecs are are chosen in uppercase, as it seems to be
>> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>> You can probably modify mod_speex to add an alias to load .spx files also.
>> 
>> -Avi
>> 
>> 
>> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> wrote:
>>> 
>>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
>>> Freeswitch and could play it with playback command. No problem. However, it
>>> is my understanding that speex format files have ".spx" extension. Why
>>> doesn't Freeswitch allows for ".spx" extension?
>>> 
>>> .speex extension are not accepted.
>>> 
>>> Thanks again
>>> 
>>> Bernard
>>> 
>>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>>> 
>>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
>>> .SPEEX (or .speex?) extension?
>>> 
>>> -Avi
>>> 
>>> 
>>> 
>>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net>
>>> wrote:
>>>> 
>>>> Hello,
>>>> 
>>>> I cannot play ".spx" (speex) file in commands such as originate?
>>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>>>> Freeswitch returns
>>>> 
>>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>>>> format [spx] for [test.spx]!
>>>> 
>>>> I converted file to wav format and could play it (speexdec test.spx
>>>> test.wav) with playback command. I probably missed a point here but I would
>>>> like to know how I can play speex files without having to convert them to
>>>> wav first.
>>>> 
>>>> Anyone?
>>>> 
>>>> Thank you
>>>> 
>>>> Bernard
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From avi at avimarcus.net  Thu Apr 19 21:35:18 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 19 Apr 2012 20:35:18 +0300
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
Message-ID: <CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>

This sounds like a totally different issue.
Please try using FreeSWITCH's encoding functionality:
http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding

-Avi


On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:

> There's still something I'm not clear with. Here's what I did
>
> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
> 2) Call a user with x-Lite soft phone: originate user/1001
> &playback(/home/sounds/a.SPEEX)
>
> ==> file is open and played but one can hear only noise squelch. What am I
> doing wrong?
>
> FYI, I then converted a.SPEEX file back to wav format and could play that
> wav file.
>
>
>
>
>
>
> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>
> > no you, cant modify it.  This is a function of mod_native_file and its
> > limited to matching the iananames they registered the codec with.
> >
> >
> >
> > On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
> >> It seems all the codecs are are chosen in uppercase, as it seems to be
> >> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
> >> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
> >> You can probably modify mod_speex to add an alias to load .spx files
> also.
> >>
> >> -Avi
> >>
> >>
> >> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
> >> wrote:
> >>>
> >>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
> >>> Freeswitch and could play it with playback command. No problem.
> However, it
> >>> is my understanding that speex format files have ".spx" extension. Why
> >>> doesn't Freeswitch allows for ".spx" extension?
> >>>
> >>> .speex extension are not accepted.
> >>>
> >>> Thanks again
> >>>
> >>> Bernard
> >>>
> >>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
> >>>
> >>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
> >>> .SPEEX (or .speex?) extension?
> >>>
> >>> -Avi
> >>>
> >>>
> >>>
> >>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net
> >
> >>> wrote:
> >>>>
> >>>> Hello,
> >>>>
> >>>> I cannot play ".spx" (speex) file in commands such as originate?
> >>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
> >>>> Freeswitch returns
> >>>>
> >>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
> >>>> format [spx] for [test.spx]!
> >>>>
> >>>> I converted file to wav format and could play it (speexdec test.spx
> >>>> test.wav) with playback command. I probably missed a point here but I
> would
> >>>> like to know how I can play speex files without having to convert
> them to
> >>>> wav first.
> >>>>
> >>>> Anyone?
> >>>>
> >>>> Thank you
> >>>>
> >>>> Bernard
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From fluixab at bellsouth.net  Thu Apr 19 22:41:29 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Thu, 19 Apr 2012 14:41:29 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
Message-ID: <CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>

/usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX

fs_encode log:
2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File [a.SPEEX] 8000hz
Frame size is 160

I got the a.SPEEX file and same squelch noise when playing it. 

2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File [a.SPEEX] 8000hz
2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms

Any specific parameters to pass to fs_encode or playback?

Thanks

Bernard

On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:

> This sounds like a totally different issue.
> Please try using FreeSWITCH's encoding functionality:
> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
> 
> -Avi
> 
> 
> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net> wrote:
> There's still something I'm not clear with. Here's what I did
> 
> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
> 2) Call a user with x-Lite soft phone: originate user/1001 &playback(/home/sounds/a.SPEEX)
> 
> ==> file is open and played but one can hear only noise squelch. What am I doing wrong?
> 
> FYI, I then converted a.SPEEX file back to wav format and could play that wav file.
> 
> 
> 
> 
> 
> 
> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
> 
> > no you, cant modify it.  This is a function of mod_native_file and its
> > limited to matching the iananames they registered the codec with.
> >
> >
> >
> > On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
> >> It seems all the codecs are are chosen in uppercase, as it seems to be
> >> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
> >> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
> >> You can probably modify mod_speex to add an alias to load .spx files also.
> >>
> >> -Avi
> >>
> >>
> >> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
> >> wrote:
> >>>
> >>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
> >>> Freeswitch and could play it with playback command. No problem. However, it
> >>> is my understanding that speex format files have ".spx" extension. Why
> >>> doesn't Freeswitch allows for ".spx" extension?
> >>>
> >>> .speex extension are not accepted.
> >>>
> >>> Thanks again
> >>>
> >>> Bernard
> >>>
> >>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
> >>>
> >>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
> >>> .SPEEX (or .speex?) extension?
> >>>
> >>> -Avi
> >>>
> >>>
> >>>
> >>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <fluixab at bellsouth.net>
> >>> wrote:
> >>>>
> >>>> Hello,
> >>>>
> >>>> I cannot play ".spx" (speex) file in commands such as originate?
> >>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
> >>>> Freeswitch returns
> >>>>
> >>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
> >>>> format [spx] for [test.spx]!
> >>>>
> >>>> I converted file to wav format and could play it (speexdec test.spx
> >>>> test.wav) with playback command. I probably missed a point here but I would
> >>>> like to know how I can play speex files without having to convert them to
> >>>> wav first.
> >>>>
> >>>> Anyone?
> >>>>
> >>>> Thank you
> >>>>
> >>>> Bernard
> >>>> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Thu Apr 19 22:58:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 13:58:44 -0500
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
	<CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
Message-ID: <CAKbxfG93BQTFDO3zyHdYgLq-CDGO=f9rq8OU3BEBRcQQM5N21Q@mail.gmail.com>

for native file you omit the extension and you cannot play it unless
the channel is also using speex.
We do not support speex files that may need to be transcoded to some
other codec.

if you have /tmp/foo.SPEEX and /tmp/foo.PCMU

you playback /tmp/foo and it picks the one to go with the native codec.
That is all we support and no other plans for speex file format
support beyond that.




On Thu, Apr 19, 2012 at 1:41 PM, Bernard Fluixa <fluixab at bellsouth.net> wrote:
> /usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX
>
> fs_encode log:
> 2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File
> [a.SPEEX] 8000hz
> Frame size is 160
>
> I got the a.SPEEX file and same squelch noise when playing it.
>
> 2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File
> [a.SPEEX] 8000hz
> 2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> Any specific parameters to pass to fs_encode or playback?
>
> Thanks
>
> Bernard
>
> On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:
>
> This sounds like a totally different issue.
> Please try using FreeSWITCH's encoding functionality:
> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>
> -Avi
>
>
> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>
> wrote:
>>
>> There's still something I'm not clear with. Here's what I did
>>
>> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
>> 2) Call a user with x-Lite soft phone: originate user/1001
>> &playback(/home/sounds/a.SPEEX)
>>
>> ==> file is open and played but one can hear only noise squelch. What am I
>> doing wrong?
>>
>> FYI, I then converted a.SPEEX file back to wav format and could play that
>> wav file.
>>
>>
>>
>>
>>
>>
>> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>>
>> > no you, cant modify it. ?This is a function of mod_native_file and its
>> > limited to matching the iananames they registered the codec with.
>> >
>> >
>> >
>> > On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> >> It seems all the codecs are are chosen in uppercase, as it seems to be
>> >> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>> >> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>> >> You can probably modify mod_speex to add an alias to load .spx files
>> >> also.
>> >>
>> >> -Avi
>> >>
>> >>
>> >> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> >> wrote:
>> >>>
>> >>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
>> >>> Freeswitch and could play it with playback command. No problem.
>> >>> However, it
>> >>> is my understanding that speex format files have ".spx" extension. Why
>> >>> doesn't Freeswitch allows for ".spx" extension?
>> >>>
>> >>> .speex extension are not accepted.
>> >>>
>> >>> Thanks again
>> >>>
>> >>> Bernard
>> >>>
>> >>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>> >>>
>> >>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
>> >>> .SPEEX (or .speex?) extension?
>> >>>
>> >>> -Avi
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa
>> >>> <fluixab at bellsouth.net>
>> >>> wrote:
>> >>>>
>> >>>> Hello,
>> >>>>
>> >>>> I cannot play ".spx" (speex) file in commands such as originate?
>> >>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>> >>>> Freeswitch returns
>> >>>>
>> >>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>> >>>> format [spx] for [test.spx]!
>> >>>>
>> >>>> I converted file to wav format and could play it (speexdec test.spx
>> >>>> test.wav) with playback command. I probably missed a point here but I
>> >>>> would
>> >>>> like to know how I can play speex files without having to convert
>> >>>> them to
>> >>>> wav first.
>> >>>>
>> >>>> Anyone?
>> >>>>
>> >>>> Thank you
>> >>>>
>> >>>> Bernard
>> >>>>
>> >>>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From bdfoster at endigotech.com  Thu Apr 19 22:59:17 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 19 Apr 2012 14:59:17 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
	<CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
Message-ID: <CAKV-jFzYPKAXZL3pQuWwvQ-8uKWa61HM=cisbiypPWkQMjKXHQ@mail.gmail.com>

   - AIFF (Apple/SGI) (extension "aiff")
   - AU (Sun/NeXT) (extension "au")
   - AVR (Audio Visual Research) (extension "avr")
   - CAF (Apple Core Audio File) (extension "caf")
   - HTK (HMM Tool Kit) (extension "htk")
   - IFF (Amiga IFF/SVX8/SV16) (extension "iff")
   - MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat")
   - MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat")
   - PAF (Ensoniq PARIS) (extension "paf")
   - PVF (Portable Voice Format) (extension "pvf")
   - RAW (header-less) (extension "raw")
   - SD2 (Sound Designer II) (extension "sd2")
   - SDS (Midi Sample Dump Standard) (extension "sds")
   - SF (Berkeley/IRCAM/CARL) (extension "sf")
   - VOC (Creative Labs) (extension "voc")
   - W64 (SoundFoundry WAVE 64) (extension "w64")
   - WAV (Microsoft) (extension "wav")
   - WAV (NIST Sphere) (extension "wav")
   - WAVEX (Microsoft) (extension "wav")
   - XI (FastTracker 2) (extension "xi")

Pick one, and live with it.


-BDF

On Thu, Apr 19, 2012 at 2:41 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:

> /usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX
>
> fs_encode log:
> 2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File
> [a.SPEEX] 8000hz
> Frame size is 160
>
> I got the a.SPEEX file and same squelch noise when playing it.
>
> 2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File
> [a.SPEEX] 8000hz
> 2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
> Any specific parameters to pass to fs_encode or playback?
>
> Thanks
>
> Bernard
>
> On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:
>
> This sounds like a totally different issue.
> Please try using FreeSWITCH's encoding functionality:
>
> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>
> -Avi
>
>
> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>wrote:
>
>> There's still something I'm not clear with. Here's what I did
>>
>> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
>> 2) Call a user with x-Lite soft phone: originate user/1001
>> &playback(/home/sounds/a.SPEEX)
>>
>> ==> file is open and played but one can hear only noise squelch. What am
>> I doing wrong?
>>
>> FYI, I then converted a.SPEEX file back to wav format and could play that
>> wav file.
>>
>>
>>
>>
>>
>>
>> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>>
>> > no you, cant modify it.  This is a function of mod_native_file and its
>> > limited to matching the iananames they registered the codec with.
>> >
>> >
>> >
>> > On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> >> It seems all the codecs are are chosen in uppercase, as it seems to be
>> >> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>> >> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>> >> You can probably modify mod_speex to add an alias to load .spx files
>> also.
>> >>
>> >> -Avi
>> >>
>> >>
>> >> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net
>> >
>> >> wrote:
>> >>>
>> >>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
>> >>> Freeswitch and could play it with playback command. No problem.
>> However, it
>> >>> is my understanding that speex format files have ".spx" extension. Why
>> >>> doesn't Freeswitch allows for ".spx" extension?
>> >>>
>> >>> .speex extension are not accepted.
>> >>>
>> >>> Thanks again
>> >>>
>> >>> Bernard
>> >>>
>> >>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>> >>>
>> >>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
>> >>> .SPEEX (or .speex?) extension?
>> >>>
>> >>> -Avi
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa <
>> fluixab at bellsouth.net>
>> >>> wrote:
>> >>>>
>> >>>> Hello,
>> >>>>
>> >>>> I cannot play ".spx" (speex) file in commands such as originate?
>> >>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>> >>>> Freeswitch returns
>> >>>>
>> >>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>> >>>> format [spx] for [test.spx]!
>> >>>>
>> >>>> I converted file to wav format and could play it (speexdec test.spx
>> >>>> test.wav) with playback command. I probably missed a point here but
>> I would
>> >>>> like to know how I can play speex files without having to convert
>> them to
>> >>>> wav first.
>> >>>>
>> >>>> Anyone?
>> >>>>
>> >>>> Thank you
>> >>>>
>> >>>> Bernard
>> >>>>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

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From fluixab at bellsouth.net  Thu Apr 19 23:10:23 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Thu, 19 Apr 2012 15:10:23 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CAKbxfG93BQTFDO3zyHdYgLq-CDGO=f9rq8OU3BEBRcQQM5N21Q@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
	<CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
	<CAKbxfG93BQTFDO3zyHdYgLq-CDGO=f9rq8OU3BEBRcQQM5N21Q@mail.gmail.com>
Message-ID: <93BB7F4B-D7B5-49E6-A1C1-987EF3C76DF3@bellsouth.net>

Anthony,

I'm clear now thanks. 

Thanks for the explanations. 

Bernard  

 

On Apr 19, 2012, at 2:58 PM, Anthony Minessale wrote:

> for native file you omit the extension and you cannot play it unless
> the channel is also using speex.
> We do not support speex files that may need to be transcoded to some
> other codec.
> 
> if you have /tmp/foo.SPEEX and /tmp/foo.PCMU
> 
> you playback /tmp/foo and it picks the one to go with the native codec.
> That is all we support and no other plans for speex file format
> support beyond that.
> 
> 
> 
> 
> On Thu, Apr 19, 2012 at 1:41 PM, Bernard Fluixa <fluixab at bellsouth.net> wrote:
>> /usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX
>> 
>> fs_encode log:
>> 2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File
>> [a.SPEEX] 8000hz
>> Frame size is 160
>> 
>> I got the a.SPEEX file and same squelch noise when playing it.
>> 
>> 2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File
>> [a.SPEEX] 8000hz
>> 2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 
>> Any specific parameters to pass to fs_encode or playback?
>> 
>> Thanks
>> 
>> Bernard
>> 
>> On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:
>> 
>> This sounds like a totally different issue.
>> Please try using FreeSWITCH's encoding functionality:
>> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>> 
>> -Avi
>> 
>> 
>> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> wrote:
>>> 
>>> There's still something I'm not clear with. Here's what I did
>>> 
>>> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
>>> 2) Call a user with x-Lite soft phone: originate user/1001
>>> &playback(/home/sounds/a.SPEEX)
>>> 
>>> ==> file is open and played but one can hear only noise squelch. What am I
>>> doing wrong?
>>> 
>>> FYI, I then converted a.SPEEX file back to wav format and could play that
>>> wav file.
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>>> 
>>>> no you, cant modify it.  This is a function of mod_native_file and its
>>>> limited to matching the iananames they registered the codec with.
>>>> 
>>>> 
>>>> 
>>>> On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>> It seems all the codecs are are chosen in uppercase, as it seems to be
>>>>> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>>>>> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>>>>> You can probably modify mod_speex to add an alias to load .spx files
>>>>> also.
>>>>> 
>>>>> -Avi
>>>>> 
>>>>> 
>>>>> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa <fluixab at bellsouth.net>
>>>>> wrote:
>>>>>> 
>>>>>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension FROM
>>>>>> Freeswitch and could play it with playback command. No problem.
>>>>>> However, it
>>>>>> is my understanding that speex format files have ".spx" extension. Why
>>>>>> doesn't Freeswitch allows for ".spx" extension?
>>>>>> 
>>>>>> .speex extension are not accepted.
>>>>>> 
>>>>>> Thanks again
>>>>>> 
>>>>>> Bernard
>>>>>> 
>>>>>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>>>>>> 
>>>>>> Didn't we answer this yesterday? Bernard, have you tried it yet with a
>>>>>> .SPEEX (or .speex?) extension?
>>>>>> 
>>>>>> -Avi
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa
>>>>>> <fluixab at bellsouth.net>
>>>>>> wrote:
>>>>>>> 
>>>>>>> Hello,
>>>>>>> 
>>>>>>> I cannot play ".spx" (speex) file in commands such as originate?
>>>>>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>>>>>>> Freeswitch returns
>>>>>>> 
>>>>>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>>>>>>> format [spx] for [test.spx]!
>>>>>>> 
>>>>>>> I converted file to wav format and could play it (speexdec test.spx
>>>>>>> test.wav) with playback command. I probably missed a point here but I
>>>>>>> would
>>>>>>> like to know how I can play speex files without having to convert
>>>>>>> them to
>>>>>>> wav first.
>>>>>>> 
>>>>>>> Anyone?
>>>>>>> 
>>>>>>> Thank you
>>>>>>> 
>>>>>>> Bernard
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Anthony Minessale II
>>>> 
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>> 
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>> 
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bdfoster at endigotech.com  Fri Apr 20 00:28:38 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 19 Apr 2012 16:28:38 -0400
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <CAMwZ-t+4TEAxTrkAMa3=fKuTBa23fzCvSRzMQPG1AH4fzwLx-g@mail.gmail.com>
References: <CB5E9EFA.2D961%krice@freeswitch.org>
	<CAMwZ-t+4TEAxTrkAMa3=fKuTBa23fzCvSRzMQPG1AH4fzwLx-g@mail.gmail.com>
Message-ID: <CAKV-jFwGO4pCbc3x_v3MWYhY+cKDw4eUnWSoUe3Zxmyd1NWbHA@mail.gmail.com>

Yea but I think the Debian guys are feeling a little left out ;-)

-BDF
On Apr 19, 2012 1:20 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:

> On Mon, Feb 13, 2012 at 10:05 AM, Ken Rice <krice at freeswitch.org> wrote:
> > Check it out... Yum is on the way for new freeswitch goodness.
>
> I have to admit that I'm loving this. It makes getting up and going so
> much better. Thanks for all the great work in this area!
>
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Fri Apr 20 00:39:40 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Thu, 19 Apr 2012 16:39:40 -0400
Subject: [Freeswitch-users] Best practice for paging in an environment with
 multiple-registrations enabled on Sofia profile
Message-ID: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>

As the subject suggests, what is the best way to do paging with
multiple-registrations enabled? In my lab I have two phones, one is
registered to 2546 and 2686, the other is registered to 2546 and 2738 (so
both phones are registered to 2546 and another extension). I have something
set up so that FS sends out a page like so:

originate {sip_auto_answer=true}user/2546 at domain.tld&playback(alertmessage.wav)

However as you might expect (it does make sense) the first phone to win the
lottery and pick up the phone first wins the call and auto answers the
call. How do I properly get every extension to playback the message at the
same time?

-BDF
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From krice at freeswitch.org  Fri Apr 20 00:44:00 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 19 Apr 2012 15:44:00 -0500
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <CAKV-jFwGO4pCbc3x_v3MWYhY+cKDw4eUnWSoUe3Zxmyd1NWbHA@mail.gmail.com>
Message-ID: <CBB5E340.314F9%krice@freeswitch.org>

The debian guys are about to get some love too... Watch out for it



On 4/19/12 3:28 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Yea but I think the Debian guys are feeling a little left out ;-)
> 
> -BDF
> 
> On Apr 19, 2012 1:20 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>> On Mon, Feb 13, 2012 at 10:05 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> > Check it out... Yum is on the way for new freeswitch goodness.
>> 
>> I have to admit that I'm loving this. It makes getting up and going so
>> much better. Thanks for all the great work in this area!
>> 
>> 
>> Best,
>> Gabe
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From abaci64 at gmail.com  Fri Apr 20 00:56:10 2012
From: abaci64 at gmail.com (Abaci)
Date: Thu, 19 Apr 2012 16:56:10 -0400
Subject: [Freeswitch-users] Looking for Kamilio - FreeSWITCH help/consulting
Message-ID: <4F907BEA.90706@gmail.com>

Hi,

We have setup a cluster of multiple FreeSWITCH servers, we are looking 
to implement Kamailio in the front as a proxy and registration server, I 
have very little experience with Kamailio, so I'm looking to hire 
someone  with knowledge and experience. please email me abaci64 at gmail 
dot com

Thanks

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From tlanghals at gmail.com  Fri Apr 20 01:00:58 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Thu, 19 Apr 2012 16:00:58 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
Message-ID: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>

I've setup a fifo park queue based off the example at
http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying to use the
Aastra BLF/Xfer key to interface with it.  The parking fifo queue works
great but I have to use two keys, one for BLF to subscribe to
queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>.

Is there any way to remove the need to have the queue+ prefix needed to
subscribe to the fifo presence so that I can just use a single BLF/Xfer key
point to <fifo_name>?

I found this thread that mentions using the map proto feature to eliminate
the need for the prefix but cannot find anything information relating to
such a feature.
http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
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From william.king at quentustech.com  Fri Apr 20 01:14:55 2012
From: william.king at quentustech.com (William King)
Date: Thu, 19 Apr 2012 14:14:55 -0700
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <CBB5E340.314F9%krice@freeswitch.org>
References: <CBB5E340.314F9%krice@freeswitch.org>
Message-ID: <4F90804F.20202@quentustech.com>

Yes, there is soon to be a decent repo for the debian based distros.
Resolving a few automation issues at the moment though.

Right now the repo will be intended to support both source packages and
binaries for:
Debian:
Stable- i386, amd64
Testing - i386, amd64

Ubuntu:
12.04- i386, amd64
10.04- i386, amd64

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/19/2012 01:44 PM, Ken Rice wrote:
> The debian guys are about to get some love too... Watch out for it
>
>
>
> On 4/19/12 3:28 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:
>
>     Yea but I think the Debian guys are feeling a little left out ;-)
>
>     -BDF
>
>     On Apr 19, 2012 1:20 PM, "Gabriel Gunderson" <gabe at gundy.org> wrote:
>
>         On Mon, Feb 13, 2012 at 10:05 AM, Ken Rice
>         <krice at freeswitch.org> wrote:
>         > Check it out... Yum is on the way for new freeswitch goodness.
>
>         I have to admit that I'm loving this. It makes getting up and
>         going so
>         much better. Thanks for all the great work in this area!
>
>
>         Best,
>         Gabe
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     ------------------------------------------------------------------------
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Fri Apr 20 02:19:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 19 Apr 2012 15:19:36 -0700
Subject: [Freeswitch-users] Looking for Kamilio - FreeSWITCH
	help/consulting
In-Reply-To: <4F907BEA.90706@gmail.com>
References: <4F907BEA.90706@gmail.com>
Message-ID: <CAKzWOxVnp5F+NN2a+60CeoYRnXO6_msybex9jmgxQfC0g3JRFQ@mail.gmail.com>

Be sure to ask on the Kamailio ML as well. I know that Daniel
Constantin-Mierla (aka "miconda", the main Kamailio author) does consulting
as do a number of others.

-MC

On Thu, Apr 19, 2012 at 1:56 PM, Abaci <abaci64 at gmail.com> wrote:

>  Hi, ****
>
> ****We have setup a cluster of multiple FreeSWITCH servers, we are
> looking to implement Kamailio in the front as a proxy and registration
> server, I have very little experience with Kamailio, so I?m looking to hire
> someone  with knowledge and experience. please email me abaci64 at gmail
> dot com
>
> ****Thanks
>
>
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From moshe3t at gmail.com  Fri Apr 20 02:22:52 2012
From: moshe3t at gmail.com (moshe3)
Date: Thu, 19 Apr 2012 18:22:52 -0400
Subject: [Freeswitch-users] Looking for Kamilio - FreeSWITCH
	help/consulting
In-Reply-To: <CAKzWOxVnp5F+NN2a+60CeoYRnXO6_msybex9jmgxQfC0g3JRFQ@mail.gmail.com>
References: <4F907BEA.90706@gmail.com>
	<CAKzWOxVnp5F+NN2a+60CeoYRnXO6_msybex9jmgxQfC0g3JRFQ@mail.gmail.com>
Message-ID: <053201cd1e7a$f676da00$e3648e00$@com>

Just a update Any SER product OpenSips or Kamailio

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, April 19, 2012 6:20 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Looking for Kamilio - FreeSWITCH
help/consulting

 

Be sure to ask on the Kamailio ML as well. I know that Daniel
Constantin-Mierla (aka "miconda", the main Kamailio author) does consulting
as do a number of others.

-MC

On Thu, Apr 19, 2012 at 1:56 PM, Abaci <abaci64 at gmail.com> wrote:

Hi, 

We have setup a cluster of multiple FreeSWITCH servers, we are looking to
implement Kamailio in the front as a proxy and registration server, I have
very little experience with Kamailio, so I'm looking to hire someone  with
knowledge and experience. please email me abaci64 at gmail dot com

Thanks 

 

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From gabe at gundy.org  Fri Apr 20 02:31:05 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 19 Apr 2012 16:31:05 -0600
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <4F90804F.20202@quentustech.com>
References: <CBB5E340.314F9%krice@freeswitch.org>
	<4F90804F.20202@quentustech.com>
Message-ID: <CAMwZ-tKiqnq3av7pqzE683p8erMF16BsPoHvBQshYKniijabaw@mail.gmail.com>

On Thu, Apr 19, 2012 at 3:14 PM, William King
<william.king at quentustech.com> wrote:
> Yes, there is soon to be a decent repo for the debian based distros.
> Resolving a few automation issues at the moment though.

William,

What can I do to be of help here?


Gabe


From william.king at quentustech.com  Fri Apr 20 02:53:31 2012
From: william.king at quentustech.com (William King)
Date: Thu, 19 Apr 2012 15:53:31 -0700
Subject: [Freeswitch-users] RPMs and Yum repo for EL5 and EL6 based
 system in beta
In-Reply-To: <CAMwZ-tKiqnq3av7pqzE683p8erMF16BsPoHvBQshYKniijabaw@mail.gmail.com>
References: <CBB5E340.314F9%krice@freeswitch.org>
	<4F90804F.20202@quentustech.com>
	<CAMwZ-tKiqnq3av7pqzE683p8erMF16BsPoHvBQshYKniijabaw@mail.gmail.com>
Message-ID: <4F90976B.8040306@quentustech.com>

At the moment there isn't anything to do. Just keep your eyes out for an
email on the mailing list asking for testers. Hopefully that email will
be soon.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/19/2012 03:31 PM, Gabriel Gunderson wrote:
> On Thu, Apr 19, 2012 at 3:14 PM, William King
> <william.king at quentustech.com> wrote:
>> Yes, there is soon to be a decent repo for the debian based distros.
>> Resolving a few automation issues at the moment though.
> William,
>
> What can I do to be of help here?
>
>
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Fri Apr 20 04:18:17 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 19:18:17 -0500
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
	dialplans
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C67BE227794@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
	<CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C67BE227794@AZWSMS03.azwarranty.int>
Message-ID: <CAKbxfG-YBy0Kyn-GuVE_pXFXvZHS2bu-s5-xKu5eGS=D1cHv+g@mail.gmail.com>

Did you miss my response?



On Wed, Apr 18, 2012 at 10:11 AM, Weigel, Stefan
<Stefan.Weigel at allianz-warranty.com> wrote:
> Hi Brian,
>
>
>
> I configured xml_curl.conf.xml with the bindings (using fs_curl as dynamic
> backend).
>
>
>
> Having this content in the static file..
>
>
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
>
>
> ? <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>
> ??? <context name="public">
>
>
>
> ???? <!--
>
> ????? <extension name="outside_call" continue="true">
>
> ??????? <condition>
>
> ????????? <action application="set" data="outside_call=true"/>
>
> ??????? </condition>
>
> ????? </extension>
>
> ???? -->
>
>
>
> ??? </context>
>
> ? </section>
>
> </document>
>
>
>
> ..the dialplan processing continues with xml curl. But as soon as any
> <action> command Is triggered, the dialplan processing doesn?t execute xml
> curl.
>
>
>
>
>
>
>
> Best regards
>
>
>
> Stefan
>
>
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian
> Foster
> Gesendet: Mittwoch, 18. April 2012 16:35
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
> dialplans
>
>
>
>
>
>>> ? <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>
>
>
> Not even close. Please read and re-read
> http://wiki.freeswitch.org/wiki/Mod_xml_curl and specifically look at
> bindings.
>
> -BDF
>
> On Wed, Apr 18, 2012 at 9:20 AM, Weigel, Stefan
> <Stefan.Weigel at allianz-warranty.com> wrote:
>
> Hi,
>
>
>
> I?m trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
> content.
>
> My static dialplan looks like:
>
>
>
>
>
> /tmp/test.xml:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
> ? <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>
> ??? <context name="public">
>
>
>
> ??? <extension name="unloop">
>
> ????? <condition field="${unroll_loops}" expression="^true$"/>
>
> ????? <condition field="${sip_looped_call}" expression="^true$">
>
> ??????? <action application="deflect" data="${destination_number}"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> ??? <extension name="outside_call" continue="true">
>
> ????? <condition>
>
> ??????? <action application="set" data="outside_call=true"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> ??? <extension name="call_debug" continue="true">
>
> ????? <condition field="${call_debug}" expression="^true$" break="never">
>
> ??????? <action application="info"/>
>
> ????? </condition>
>
> ??? </extension>
>
>
>
> ??? </context>
>
> ? </section>
>
> </document>
>
>
>
> I changed parameter ?dialplan? in profile internal.xml:
>
>
>
> ??? <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>
>
>
> But when placing a call I only see that diaplan from static XML is parsed
> and executed (message ?has executed the last dialplan instruction, hanging
> up.?).
>
>
>
>
>
> What?s going wrong ?
>
>
>
>
>
>
>
> Thanks and cheers
>
>
>
> Stefan
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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MSN:anthony_minessale at hotmail.com
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FreeSWITCH Developer Conference
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pstn:+19193869900


From anthony.minessale at gmail.com  Fri Apr 20 04:24:21 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 19:24:21 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
Message-ID: <CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>

look in tree for the installed presence_map.conf.xml


its a series of ext range regex vs protos


you must also set the sofia profile param presence-proto-lookup to true



On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals <tlanghals at gmail.com> wrote:
> I've setup a fifo park queue based off the example at
> http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying to use the
> Aastra BLF/Xfer key to interface with it.? The parking fifo queue works
> great but I have to use two keys, one for BLF to subscribe to
> queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>.
>
> Is there any way to remove the need to have the queue+ prefix needed to
> subscribe to the fifo presence so that I can just use a single BLF/Xfer key
> point to <fifo_name>?
>
> I found this thread that mentions using the map proto feature to eliminate
> the need for the prefix but cannot find anything information relating to
> such a feature.
> http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Apr 20 04:37:47 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 19 Apr 2012 19:37:47 -0500
Subject: [Freeswitch-users] Originate call into conference - no early
	media
In-Reply-To: <4F8F0678.6060906@integrafin.co.uk>
References: <4F8F0678.6060906@integrafin.co.uk>
Message-ID: <CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>

You only get it when you actually are being sent early media.
Probably the ones that dont have any sound are only sending 180 so
there is nothing you can do about it.



On Wed, Apr 18, 2012 at 1:22 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
> Hi,
>
> I'm using the following dialplan snippet to call new parties into a
> conference (see :
> http://wiki.freeswitch.org/wiki/Conference_Add_Call_Example):
>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}user/$1 &conference(${conference_name})}"/>
> </condition>
>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1
> &conference(${conference_name})}"/>
> </condition>
>
>
>
>
> The page referenced implies I should get early media when I call someone
> into the conference, but when I try (either for local endpoints or those
> on a remote server via a gateway) I don't hear any ringing. I do if I
> call via a PSTN gateway.
>
> Any ideas?
>
> Cheers
>
> Alex
>
> --
> This message is intended only for the addressee and may contain
> confidential information. ?Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
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>
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> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
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FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From tlanghals at gmail.com  Fri Apr 20 06:20:31 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Thu, 19 Apr 2012 21:20:31 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
Message-ID: <CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>

I added <param name="presence-proto-lookup" value="true"/> to my sip
interface config and modified presence_map.conf.xml as follows but still no
luck.  I tried proto="queue", proto="queue+", proto="fifo" and restarted
freeswitch between each test.  My fifos are named park_slot1, park_slot2,
etc.

<configuration name="presence_map.conf" description="PRESENCE MAP">
  <domains>
    <domain name="$${domain}">
      <exten regex="park_slot\d+" proto="queue"/>
    </domain>
  </domains>
</configuration>


On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> look in tree for the installed presence_map.conf.xml
>
>
> its a series of ext range regex vs protos
>
>
> you must also set the sofia profile param presence-proto-lookup to true
>
>
>
> On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals <tlanghals at gmail.com>
> wrote:
> > I've setup a fifo park queue based off the example at
> > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying to use
> the
> > Aastra BLF/Xfer key to interface with it.  The parking fifo queue works
> > great but I have to use two keys, one for BLF to subscribe to
> > queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>.
> >
> > Is there any way to remove the need to have the queue+ prefix needed to
> > subscribe to the fifo presence so that I can just use a single BLF/Xfer
> key
> > point to <fifo_name>?
> >
> > I found this thread that mentions using the map proto feature to
> eliminate
> > the need for the prefix but cannot find anything information relating to
> > such a feature.
> >
> http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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From jmsucasas at me.com  Fri Apr 20 04:36:17 2012
From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto)
Date: Fri, 20 Apr 2012 02:36:17 +0200
Subject: [Freeswitch-users] Packet-time negotiation problem
Message-ID: <C36BAED3-144F-44C7-9F8E-1CBF50BCAD5C@me.com>

Hello,

	I have installed freeSwitch 1.1beta from the git repository (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000). I have enabled iLBC at 30i codec by setting global vars:

	<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,iLBC at 30i,SILK,G729"/>
	<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,iLBC at 30i,SILK,G729"/>

	I try to call from Bria for iPhone to x-Lite 4.1.  Bria offers iLBC at 30i:

2012-04-20 00:24:03.740477 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1001 at sukmad.dyndns-home.com [624007f0-d580-4500-834f-76c2ba578df0]
2012-04-20 00:24:03.740477 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1001 at sukmad.dyndns-home.com) Running State Change CS_NEW
2012-04-20 00:24:03.740477 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1001 at sukmad.dyndns-home.com) State NEW
2012-04-20 00:24:03.740477 [DEBUG] sofia.c:5561 Channel sofia/internal/1001 at sukmad.dyndns-home.com entering state [received][100]
2012-04-20 00:24:03.740477 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=- 3543863125 3543863125 IN IP4 88.9.239.104
s=cpc_med
c=IN IP4 88.9.239.104
t=0 0
m=audio 17002 RTP/AVP 112 18 105 101
c=IN IP4 88.9.239.104
a=rtpmap:112 SILK/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

	But FS set the codec to iLBC 98 (according to vars.xml,  it corresponds to iLBC at 20i):

2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G7221:115:32000:20:48000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G7221:115:32000:20:48000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G7221:107:16000:20:32000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G7221:107:16000:20:32000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[G722:9:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[G722:9:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[PCMU:0:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[PCMU:0:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[PCMA:8:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[PCMA:8:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[GSM:3:8000:20:13200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[GSM:3:8000:20:13200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[GSM:3:8000:20:13200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [SILK:112:8000:20:0]/[iLBC:98:8000:20:15200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:98:8000:20:15200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [iLBC:105:8000:30:0]/[iLBC:98:8000:20:15200]
2012-04-20 00:24:03.740477 [DEBUG] sofia_glue.c:3006 Set Codec sofia/internal/1001 at sukmad.dyndns-home.com iLBC/8000 20 ms 160 samples 15200 bits
2012-04-20 00:24:03.740477 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at sukmad.dyndns-home.com Original read codec set to iLBC:98

	I'm trying to fix the codec in the dialplan by setting absolute_codec_string channel variable:

Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action bridge({absolute_codec_string='iLBC at 30i,G729'}user/${dialed_extension}@${domain_name}) 
Dialplan: sofia/internal/1001 at sukmad.dyndns-home.com Action answer() 

	When FS rings in the B-leg,  I think it sends an incorrect codec (mode=20)

2012-04-20 00:24:04.170478 [DEBUG] mod_sofia.c:2582 Ring SDP:
v=0
o=FreeSWITCH 1334863010 1334863011 IN IP4 88.9.239.104
s=FreeSWITCH
c=IN IP4 88.9.239.104
t=0 0
m=audio 11234 RTP/AVP 105 101
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

	When I try to answer in B-leg (x-Lite softphone) it offers iLBC 98 (30 ms):

2012-04-20 00:24:13.750480 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/sip:1002 at 192.168.0.2:3348 [BREAK]
2012-04-20 00:24:13.750480 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/sip:1002 at 192.168.0.2:3348 [BREAK]
2012-04-20 00:24:13.750480 [DEBUG] sofia.c:5561 Channel sofia/internal/sip:1002 at 192.168.0.2:3348 entering state [completing][200]
2012-04-20 00:24:13.750480 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=- 1334874336856091 1 IN IP4 192.168.1.2
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.2
t=0 0
m=audio 54624 RTP/AVP 98 101
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

	and the call ends with the following error:

2012-04-20 00:24:13.800478 [DEBUG] switch_core_session.c:793 Send signal sofia/internal/1001 at sukmad.dyndns-home.com [BREAK]
2012-04-20 00:24:13.810479 [DEBUG] switch_core_io.c:1095 Engaging Write Buffer at 320 bytes to accommodate 480->320
2012-04-20 00:24:13.980477 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 20 to 30
2012-04-20 00:24:13.980477 [DEBUG] sofia_glue.c:2924 Changing Codec from iLBC at 20ms@8000hz to iLBC at 30ms@8000hz
2012-04-20 00:24:14.020479 [WARNING] switch_core_codec.c:691 Codec iLBC Exists but not at the desired implementation. 8000hz 30ms
2012-04-20 00:24:14.020479 [ERR] sofia_glue.c:2954 Can't load codec?
2012-04-20 00:24:14.020479 [DEBUG] switch_channel.c:2848 (sofia/internal/1001 at sukmad.dyndns-home.com) Callstate Change ACTIVE -> HANGUP
2012-04-20 00:24:14.020479 [NOTICE] sofia_glue.c:2955 Hangup sofia/internal/1001 at sukmad.dyndns-home.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

	Is it possible to re-packetize iLBC streams from iLBC at 20i to iLBC at 30i?.  Why FS choose an incorrect packet time?. 
	I have read about this problem in Internet and I try to uncomment the following line:

	<param name="rtp-autofix-timing" value="false"/>

	But when I try to make the call I don't hear nothing (except a big noise)...  There are any solution?.  I attach all the dump from fs_cli.


Thanks in advance.


Regards.







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From acrow at integrafin.co.uk  Fri Apr 20 12:07:14 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 20 Apr 2012 09:07:14 +0100
Subject: [Freeswitch-users] Originate call into conference - no early
 media
In-Reply-To: <CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
Message-ID: <4F911932.7030002@integrafin.co.uk>

On 20/04/12 01:37, Anthony Minessale wrote:
> You only get it when you actually are being sent early media.
> Probably the ones that dont have any sound are only sending 180 so
> there is nothing you can do about it.
>

Anthony,

Thanks - would this only be "unfixable" for internal phones, as I'm sure 
I could change at least my other freeswitch boxes I am gatewaying to to 
send early media?

Cheers

Alex



From barnyritchley at hotmail.com  Fri Apr 20 12:14:47 2012
From: barnyritchley at hotmail.com (Barnaby Ritchley)
Date: Fri, 20 Apr 2012 09:14:47 +0100
Subject: [Freeswitch-users] SIP 183 on a leg before bleg invite in bridge
Message-ID: <BLU0-SMTP15737FA192AEB524E8C7BC2BD220@phx.gbl>

Hi Guys

I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg.

Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing.

The call goes:

Cisco UA (1.1.1.1)  -> FS (2.2.2.2) -> Gateway (3.3.3.3)

Sofia config has nothing special in it, relevent sections:

<param name="inbound-codec-negotiation" value="generous"/>
<param name="disable-transcoding" value="true"/>
<param name="manual-redirect" value="true"/>
<param name="disable-transfer" value="true"/>

and the dial plan is straight forward, just a bridge between the two endpoints.

Any ideas?




Here is the trace of the call:

INVITE sip:1234 at 2.2.2.2:5060 SIP/2.0.
Date: Fri, 20 Apr 2012 07:58:36 GMT.
Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500".
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
Allow-Events: presence, kpml.
Supported: 100rel,timer,resource-priority,replaces.
Supported: X-cisco-srtp-fallback.
Supported: Geolocation.
Min-SE:  1800.
Content-Length: 212.
To: <sip:1234 at 2.2.2.2>.
Expires: 180.
Content-Type: application/sdp.
Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
CSeq: 101 INVITE.
Session-Expires:  1800.
Max-Forwards: 69.
.
v=0.
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1.
s=SIP Call.
c=IN IP4 1.1.1.1.
t=0 0.
m=audio 31422 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.


U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
To: <sip:1234 at 2.2.2.2>.
CSeq: 101 INVITE.
User-Agent: FreeSWITCH.
Content-Length: 0.
.


U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
To: <sip:1234 at 2.2.2.2>;tag=vm45yDZjgmDKH.
Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
CSeq: 101 INVITE.
Contact: <sip:1234 at 2.2.2.2:5060;transport=udp>.
User-Agent: FreeSWITCH.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 247.
Remote-Party-ID: "1234" <sip:1234 at 2.2.2.2>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2.
s=FreeSWITCH.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 26820 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.


U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060
INVITE sip:05600660128 at 3.3.3.3:5060 SIP/2.0.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS.
Max-Forwards: 68.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
Contact: <sip:gw+C4DF4B2A088C91 at 2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>.
User-Agent: FreeSWITCH.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 201.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "00000000" <sip:00000000 at 2.2.2.2>;party=calling;screen=no;privacy=full.
.
v=0.
o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2.
s=FreeSWITCH.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 21466 RTP/AVP 8 101 13.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.


U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
User-Agent: SBC01.
Content-Length: 0.
.


U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
To: <sip:1234 at 3.3.3.3:5060>;tag=7NNjeSyyjtHSB.
Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
CSeq: 27118284 INVITE.
Contact: <sip:05600660128 at 3.3.3.3:5060;transport=udp>.
User-Agent: SBC01.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, hold, refer.
Content-Length: 0.



From miha at softnet.si  Fri Apr 20 12:14:54 2012
From: miha at softnet.si (Miha)
Date: Fri, 20 Apr 2012 10:14:54 +0200
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <CANBhyPBfXzDo3f0x7VOUw+9nc0sKrGxQSMwc_LJRCC-7b1MumA@mail.gmail.com>
References: <4F8D3AB1.1030207@softnet.si>
	<CANBhyPBfXzDo3f0x7VOUw+9nc0sKrGxQSMwc_LJRCC-7b1MumA@mail.gmail.com>
Message-ID: <4F911AFE.2@softnet.si>

On 4/17/2012 2:15 PM, Anton Kvashenkin wrote:
> Good read 
> http://blog.2600hz.com/post/18096749028/understanding-how-fs-supports-opensips-as-a 
>
>
> 17 ?????? 2012 ?. 13:41 ???????????? Miha <miha at softnet.si 
> <mailto:miha at softnet.si>> ???????:
>
>     Hi to all,
>
>     I have configured opensips that works like load_balancer (wiki on
>     FS page).
>     So, we have SBC, trunk is made to Opensips, than opensips is
>     load_balacing to FSs.
>     When you call from FS, call is going to Opensips, then opensips
>     redirect the call to SBC, if call must go outside.
>
>     My question:
>
>     FSs boxes are all the same (same dialplan, directory
>     configuration, etc.). What happens if call is made to a user,
>     which is registered on one of the FS box? I guess I must change
>     dialplan and set that all trafic FS send to opensips (not
>     internally), because if someone calls to a local user, but user is
>     registered on other FS box, FS will behave as user is not registered.
>
>     So, will opensips know which user is registered on which FS box
>     when opensips will be doing load_balacing?
>
>     Thanks for explenation.
>
>     Regards,
>     Miha
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
HI,

thank you all for your answers . So If I would like to do load_balacing 
with opensips I would need some kind of logic for every registered phone 
I must know to which FS is registerd, so I would know to which one I 
must FDW call.

But in this case this script do not work (as is written on FS wiki):

  if (is_method("INVITE")) {
                 if (!load_balance("1","pstn","1")) {
                         send_reply("503","Service Unavailable");
                         exit;
                 }
         }
         else if (is_method("REGISTER")) {
                 if (!ds_select_dst("1", "0")) {
                         send_reply("503","Service Unavailable");
                         exit;
                 }
         }
         else {
                 send_reply("405","Method Not Allowed");
                 exit;
         }

         # route the request
         if (!t_relay()) {
                 sl_reply_error();
         }




Example:. You have three users.A and B user are registered to FS1 and C user to FS2. A and C are talking if B gets a call as Opensips do not know to which box B is registered

It could FWD it to FS2, which will not work as B is registered to FS1. 
Am I right?

Rergards,
Miha
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From barnyritchley at hotmail.com  Fri Apr 20 12:45:46 2012
From: barnyritchley at hotmail.com (Barnaby Ritchley)
Date: Fri, 20 Apr 2012 09:45:46 +0100
Subject: [Freeswitch-users] SIP 183 on a leg before bleg invite in bridge
In-Reply-To: <BLU0-SMTP15737FA192AEB524E8C7BC2BD220@phx.gbl>
References: <BLU0-SMTP15737FA192AEB524E8C7BC2BD220@phx.gbl>
Message-ID: <BLU0-SMTP2683E299ECF4D1A5483C80EBD220@phx.gbl>

Answered my own question...


the problem occurrs when using call recording on a channel.  With the dial plan command:

<action application="record_session" data="somefile.wav"/>

the problem is experienced.  If we set call recording to only happen on the B-Leg using bridge:

<action application="bridge" data="{execute_on_answer=record_session 'somefile.wav'}sofia/gateway/gateway/1234"/>

then the problem doesnt exist.

I guess this is because FS is settling on a codec in order to start call recording, so the only way around this would be for FS to generate the ringing.  Im sure there is a way to do this, but in my particular setup this is not required, and setting the recording on the b-leg works just fine.

Many thanks

On 20 Apr 2012, at 09:14, Barnaby Ritchley wrote:

> Hi Guys
> 
> I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg.
> 
> Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing.
> 
> The call goes:
> 
> Cisco UA (1.1.1.1)  -> FS (2.2.2.2) -> Gateway (3.3.3.3)
> 
> Sofia config has nothing special in it, relevent sections:
> 
> <param name="inbound-codec-negotiation" value="generous"/>
> <param name="disable-transcoding" value="true"/>
> <param name="manual-redirect" value="true"/>
> <param name="disable-transfer" value="true"/>
> 
> and the dial plan is straight forward, just a bridge between the two endpoints.
> 
> Any ideas?
> 
> 
> 
> 
> Here is the trace of the call:
> 
> INVITE sip:1234 at 2.2.2.2:5060 SIP/2.0.
> Date: Fri, 20 Apr 2012 07:58:36 GMT.
> Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500".
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY.
> From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
> Allow-Events: presence, kpml.
> Supported: 100rel,timer,resource-priority,replaces.
> Supported: X-cisco-srtp-fallback.
> Supported: Geolocation.
> Min-SE:  1800.
> Content-Length: 212.
> To: <sip:1234 at 2.2.2.2>.
> Expires: 180.
> Content-Type: application/sdp.
> Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
> CSeq: 101 INVITE.
> Session-Expires:  1800.
> Max-Forwards: 69.
> .
> v=0.
> o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1.
> s=SIP Call.
> c=IN IP4 1.1.1.1.
> t=0 0.
> m=audio 31422 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=ptime:20.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> 
> 
> U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
> From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
> To: <sip:1234 at 2.2.2.2>.
> CSeq: 101 INVITE.
> User-Agent: FreeSWITCH.
> Content-Length: 0.
> .
> 
> 
> U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979.
> From: "User" <sip:00000000 at 1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567.
> To: <sip:1234 at 2.2.2.2>;tag=vm45yDZjgmDKH.
> Call-ID: a088a380-f911172c-3d33e6-8d200795 at 1.1.1.1.
> CSeq: 101 INVITE.
> Contact: <sip:1234 at 2.2.2.2:5060;transport=udp>.
> User-Agent: FreeSWITCH.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 247.
> Remote-Party-ID: "1234" <sip:1234 at 2.2.2.2>;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2.
> s=FreeSWITCH.
> c=IN IP4 2.2.2.2.
> t=0 0.
> m=audio 26820 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> 
> 
> U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060
> INVITE sip:05600660128 at 3.3.3.3:5060 SIP/2.0.
> Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS.
> Max-Forwards: 68.
> From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
> To: <sip:1234 at 3.3.3.3:5060>.
> Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
> CSeq: 27118284 INVITE.
> Contact: <sip:gw+C4DF4B2A088C91 at 2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>.
> User-Agent: FreeSWITCH.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 201.
> X-FS-Support: update_display,send_info.
> Remote-Party-ID: "00000000" <sip:00000000 at 2.2.2.2>;party=calling;screen=no;privacy=full.
> .
> v=0.
> o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2.
> s=FreeSWITCH.
> c=IN IP4 2.2.2.2.
> t=0 0.
> m=audio 21466 RTP/AVP 8 101 13.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> 
> 
> U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
> From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
> To: <sip:1234 at 3.3.3.3:5060>.
> Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
> CSeq: 27118284 INVITE.
> User-Agent: SBC01.
> Content-Length: 0.
> .
> 
> 
> U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS.
> From: "00000000" <sip:00000000 at 2.2.2.2>;tag=4vjXj69rtpaeS.
> To: <sip:1234 at 3.3.3.3:5060>;tag=7NNjeSyyjtHSB.
> Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b.
> CSeq: 27118284 INVITE.
> Contact: <sip:05600660128 at 3.3.3.3:5060;transport=udp>.
> User-Agent: SBC01.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, hold, refer.
> Content-Length: 0.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From khorsmann at gmail.com  Fri Apr 20 12:56:02 2012
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Fri, 20 Apr 2012 10:56:02 +0200
Subject: [Freeswitch-users] FS as SBC gets USER_NOT_REGISTERED with
	registered Users from external
Message-ID: <CAFArqsbSRuxPZ-nJ81ZDSAqGxvX3gsHxLJwiMU1BdB_3tsDgrw@mail.gmail.com>

Hi List,

I add the register stuff to the public.xml dialplan to force external
users to register.
i have problems to call registered users. Why i get USER_NOT_REGISTERED?

I make an siptrace within Freeswitch and put it here:
http://pastebin.freeswitch.org/18893

Thanks.


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*


From wstephen80 at gmail.com  Fri Apr 20 19:34:28 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 20 Apr 2012 17:34:28 +0200
Subject: [Freeswitch-users] annexb=yes
Message-ID: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>

In my Freeswitch, the originated outgoing calls have:

Media Attribute (a): fmtp:18 annexb=no

How can I change it in annexb=yes ?

Stephen
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From msc at freeswitch.org  Fri Apr 20 19:46:19 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 Apr 2012 08:46:19 -0700
Subject: [Freeswitch-users] annexb=yes
In-Reply-To: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>
References: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>
Message-ID: <CAKzWOxX+x3A79S8vdgQkCRip=BM6PyUmEc02J_=KPxs2-APZkQ@mail.gmail.com>

What device is originating the calls?
-MC

On Fri, Apr 20, 2012 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> In my Freeswitch, the originated outgoing calls have:
>
> Media Attribute (a): fmtp:18 annexb=no
>
> How can I change it in annexb=yes ?
>
> Stephen
>
>
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From gmaruzz at gmail.com  Fri Apr 20 19:58:34 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 20 Apr 2012 17:58:34 +0200
Subject: [Freeswitch-users] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
Message-ID: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>

Hello FreeSWITCHers,

the new version of GSMopen is basically usable, and the docs at the
wiki almost up to date (barring the details of the config file, you
don't need them now :) ).

Good news is that GSMopen now supports Huawei E1550 and compatible
dongles, that are cheap (roughly $20 on Ebay) and reliable.

Thanks to dongle's pcm serial streaming via USB, no more need for
sound hacks, ALSA, or any sound subsystem (and cpu load). All is
digital.

It's reasonable to expect to be able to support many dozen concurrent
calls per machine (dongles are USB 2.0 compliant).

Works well right now on Linux and Windows, and would probably works on
*BSD and OSX with minor tweaks.

Code will be polished, refactured, rewritten, etc. But the thing is there.

So, please update your source code to latest git, have a look at the
wiki page, build mod_gsmopen, connect a dongle to your machine, and
begin catching bugs and documentation inaccuracies. And asking for
features for your use cases.

Thanks to you all in advance, it's a pleasure and an honor to be part
of this community.

http://wiki.freeswitch.org/wiki/Gsmopen

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From fs-list at communicatefreely.net  Fri Apr 20 20:04:56 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 20 Apr 2012 12:04:56 -0400
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
 queue?
In-Reply-To: <CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
	<CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
Message-ID: <4F918928.5080608@communicatefreely.net>

I'm also using the Aastra phones, and I had to use the presence map to 
make things work.

proto="queue" is correct.
My regex is a bit different, as my parking slots have slightly different 
naming conventions, but it works.

You may want to look at the subscriptions to make sure the domains 
match.  Are you using ODBC for sofia?  If so, you can look in the 
sip_subscriptions table and see the domain that was subscribed to.  That 
domain has to be the same as the domain you use in the presence map.

Also in your sofia profile, there is an option to force subscription 
domain.  Are you using that?  You don't have to if the domains match.

With the Aastra phones, the domain used in the subscribe and also in 
invites is whatever you put in sip proxy and sip registrar (still 
haven't figured out which is which).  Are you using your SIP domain, or 
an IP address?  If you are using an IP address, make the proxy and 
registrar fields the SIP domain, and put the IP address and port in the 
outbound proxy section.

Hope that's helpful.

-Tim

Travis Langhals wrote:
> I added <param name="presence-proto-lookup" value="true"/> to my sip 
> interface config and modified presence_map.conf.xml as follows but 
> still no luck.  I tried proto="queue", proto="queue+", proto="fifo" 
> and restarted freeswitch between each test.  My fifos are named 
> park_slot1, park_slot2, etc.
>
> <configuration name="presence_map.conf" description="PRESENCE MAP">
>   <domains>
>     <domain name="$${domain}">
>       <exten regex="park_slot\d+" proto="queue"/>
>     </domain>
>   </domains>
> </configuration>
>
>
> On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     look in tree for the installed presence_map.conf.xml
>
>
>     its a series of ext range regex vs protos
>
>
>     you must also set the sofia profile param presence-proto-lookup to
>     true
>
>
>
>     On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals
>     <tlanghals at gmail.com <mailto:tlanghals at gmail.com>> wrote:
>     > I've setup a fifo park queue based off the example at
>     > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying
>     to use the
>     > Aastra BLF/Xfer key to interface with it.  The parking fifo
>     queue works
>     > great but I have to use two keys, one for BLF to subscribe to
>     > queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>.
>     >
>     > Is there any way to remove the need to have the queue+ prefix
>     needed to
>     > subscribe to the fifo presence so that I can just use a single
>     BLF/Xfer key
>     > point to <fifo_name>?
>     >
>     > I found this thread that mentions using the map proto feature to
>     eliminate
>     > the need for the prefix but cannot find anything information
>     relating to
>     > such a feature.
>     >
>     http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>
>     --
>     Anthony Minessale II
>
>     FreeSWITCH http://www.freeswitch.org/
>     ClueCon http://www.cluecon.com/
>     Twitter: http://twitter.com/FreeSWITCH_wire
>
>     AIM: anthm
>     MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:+19193869900 <tel:%2B19193869900>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From fs-list at communicatefreely.net  Fri Apr 20 20:06:53 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 20 Apr 2012 12:06:53 -0400
Subject: [Freeswitch-users] Best practice for paging in an environment
 with multiple-registrations enabled on Sofia profile
In-Reply-To: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
References: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
Message-ID: <4F91899D.9020104@communicatefreely.net>

Not to change the subject, but is multi-cast an option for you?  That 
solves a lot of problems, and Freeswitch is capable of generating the 
multi-cast stream.

-Tim

Brian Foster wrote:
>
> As the subject suggests, what is the best way to do paging with 
> multiple-registrations enabled? In my lab I have two phones, one is 
> registered to 2546 and 2686, the other is registered to 2546 and 2738 
> (so both phones are registered to 2546 and another extension). I have 
> something set up so that FS sends out a page like so:
>
> originate {sip_auto_answer=true}user/2546 at domain.tld 
> &playback(alertmessage.wav)
>
> However as you might expect (it does make sense) the first phone to 
> win the lottery and pick up the phone first wins the call and auto 
> answers the call. How do I properly get every extension to playback 
> the message at the same time?
>
> -BDF
>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From wstephen80 at gmail.com  Fri Apr 20 20:10:12 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 20 Apr 2012 18:10:12 +0200
Subject: [Freeswitch-users] annexb=yes
In-Reply-To: <CAKzWOxX+x3A79S8vdgQkCRip=BM6PyUmEc02J_=KPxs2-APZkQ@mail.gmail.com>
References: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>
	<CAKzWOxX+x3A79S8vdgQkCRip=BM6PyUmEc02J_=KPxs2-APZkQ@mail.gmail.com>
Message-ID: <CAO4LeU2a2o=PztcG_HCm2RBbi4woEB5OnKVs5eN=cei_4PcmGw@mail.gmail.com>

The call is coming from different providers and I do a bridge to originate
the B leg.
I want to force in this outgoing leg the annexb=yes: it's possible to force
it int outgoing call leg?

On Fri, Apr 20, 2012 at 5:46 PM, Michael Collins <msc at freeswitch.org> wrote:

> What device is originating the calls?
> -MC
>
> On Fri, Apr 20, 2012 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> In my Freeswitch, the originated outgoing calls have:
>>
>> Media Attribute (a): fmtp:18 annexb=no
>>
>> How can I change it in annexb=yes ?
>>
>> Stephen
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From djbinter at gmail.com  Fri Apr 20 20:13:34 2012
From: djbinter at gmail.com (DJB International)
Date: Fri, 20 Apr 2012 09:13:34 -0700
Subject: [Freeswitch-users] annexb=yes
In-Reply-To: <CAO4LeU2a2o=PztcG_HCm2RBbi4woEB5OnKVs5eN=cei_4PcmGw@mail.gmail.com>
References: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>
	<CAKzWOxX+x3A79S8vdgQkCRip=BM6PyUmEc02J_=KPxs2-APZkQ@mail.gmail.com>
	<CAO4LeU2a2o=PztcG_HCm2RBbi4woEB5OnKVs5eN=cei_4PcmGw@mail.gmail.com>
Message-ID: <CAAB3jWvK-Kh2Uj4kFt7TOZ92y9N1a9quV_-Az915Q_MR3exQ4g@mail.gmail.com>

<action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=yes"/>



On Fri, Apr 20, 2012 at 9:10 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> The call is coming from different providers and I do a bridge to originate
> the B leg.
> I want to force in this outgoing leg the annexb=yes: it's possible to
> force it int outgoing call leg?
>
> On Fri, Apr 20, 2012 at 5:46 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> What device is originating the calls?
>> -MC
>>
>> On Fri, Apr 20, 2012 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> In my Freeswitch, the originated outgoing calls have:
>>>
>>> Media Attribute (a): fmtp:18 annexb=no
>>>
>>> How can I change it in annexb=yes ?
>>>
>>> Stephen
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Fri Apr 20 20:20:40 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 20 Apr 2012 11:20:40 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <4F918928.5080608@communicatefreely.net>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
	<CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
	<4F918928.5080608@communicatefreely.net>
Message-ID: <CAKbxfG_H_=maHQ67s7EEY5-CsFf4QJJryFmDJn1vTHdkXd2cCw@mail.gmail.com>

You might want to stick with the domain=* unless you know the hosts in
your sip packet match the $${domain} for sure.
it's "queue" for fifo and you need to make sure the regex matches any
extensions in question that you are subscribing to.

On Fri, Apr 20, 2012 at 11:04 AM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> I'm also using the Aastra phones, and I had to use the presence map to
> make things work.
>
> proto="queue" is correct.
> My regex is a bit different, as my parking slots have slightly different
> naming conventions, but it works.
>
> You may want to look at the subscriptions to make sure the domains
> match. ?Are you using ODBC for sofia? ?If so, you can look in the
> sip_subscriptions table and see the domain that was subscribed to. ?That
> domain has to be the same as the domain you use in the presence map.
>
> Also in your sofia profile, there is an option to force subscription
> domain. ?Are you using that? ?You don't have to if the domains match.
>
> With the Aastra phones, the domain used in the subscribe and also in
> invites is whatever you put in sip proxy and sip registrar (still
> haven't figured out which is which). ?Are you using your SIP domain, or
> an IP address? ?If you are using an IP address, make the proxy and
> registrar fields the SIP domain, and put the IP address and port in the
> outbound proxy section.
>
> Hope that's helpful.
>
> -Tim
>
> Travis Langhals wrote:
>> I added <param name="presence-proto-lookup" value="true"/> to my sip
>> interface config and modified presence_map.conf.xml as follows but
>> still no luck. ?I tried proto="queue", proto="queue+", proto="fifo"
>> and restarted freeswitch between each test. ?My fifos are named
>> park_slot1, park_slot2, etc.
>>
>> <configuration name="presence_map.conf" description="PRESENCE MAP">
>> ? <domains>
>> ? ? <domain name="$${domain}">
>> ? ? ? <exten regex="park_slot\d+" proto="queue"/>
>> ? ? </domain>
>> ? </domains>
>> </configuration>
>>
>>
>> On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale
>> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>>
>> ? ? look in tree for the installed presence_map.conf.xml
>>
>>
>> ? ? its a series of ext range regex vs protos
>>
>>
>> ? ? you must also set the sofia profile param presence-proto-lookup to
>> ? ? true
>>
>>
>>
>> ? ? On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals
>> ? ? <tlanghals at gmail.com <mailto:tlanghals at gmail.com>> wrote:
>> ? ? > I've setup a fifo park queue based off the example at
>> ? ? > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying
>> ? ? to use the
>> ? ? > Aastra BLF/Xfer key to interface with it. ?The parking fifo
>> ? ? queue works
>> ? ? > great but I have to use two keys, one for BLF to subscribe to
>> ? ? > queue+<fifo_name> and another for speed dial/tranfer to <fifo_name>.
>> ? ? >
>> ? ? > Is there any way to remove the need to have the queue+ prefix
>> ? ? needed to
>> ? ? > subscribe to the fifo presence so that I can just use a single
>> ? ? BLF/Xfer key
>> ? ? > point to <fifo_name>?
>> ? ? >
>> ? ? > I found this thread that mentions using the map proto feature to
>> ? ? eliminate
>> ? ? > the need for the prefix but cannot find anything information
>> ? ? relating to
>> ? ? > such a feature.
>> ? ? >
>> ? ? http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>> ? ? >
>> ? ? >
>> ? ? _________________________________________________________________________
>> ? ? > Professional FreeSWITCH Consulting Services:
>> ? ? > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> ? ? > http://www.freeswitchsolutions.com
>> ? ? >
>> ? ? > 
>> ? ? > 
>> ? ? >
>> ? ? > Official FreeSWITCH Sites
>> ? ? > http://www.freeswitch.org
>> ? ? > http://wiki.freeswitch.org
>> ? ? > http://www.cluecon.com
>> ? ? >
>> ? ? > FreeSWITCH-users mailing list
>> ? ? > FreeSWITCH-users at lists.freeswitch.org
>> ? ? <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? >
>> ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? > http://www.freeswitch.org
>> ? ? >
>>
>>
>>
>> ? ? --
>> ? ? Anthony Minessale II
>>
>> ? ? FreeSWITCH http://www.freeswitch.org/
>> ? ? ClueCon http://www.cluecon.com/
>> ? ? Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> ? ? AIM: anthm
>> ? ? MSN:anthony_minessale at hotmail.com
>> ? ? <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> ? ? GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> ? ? <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> ? ? IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> ? ? FreeSWITCH Developer Conference
>> ? ? sip:888 at conference.freeswitch.org
>> ? ? <mailto:sip%3A888 at conference.freeswitch.org>
>> ? ? googletalk:conf+888 at conference.freeswitch.org
>> ? ? <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> ? ? pstn:+19193869900 <tel:%2B19193869900>
>>
>> ? ? _________________________________________________________________________
>> ? ? Professional FreeSWITCH Consulting Services:
>> ? ? consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> ? ? http://www.freeswitchsolutions.com
>>
>> ? ? 
>> ? ? 
>>
>> ? ? Official FreeSWITCH Sites
>> ? ? http://www.freeswitch.org
>> ? ? http://wiki.freeswitch.org
>> ? ? http://www.cluecon.com
>>
>> ? ? FreeSWITCH-users mailing list
>> ? ? FreeSWITCH-users at lists.freeswitch.org
>> ? ? <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> ? ? http://www.freeswitch.org
>>
>>
>> ------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From wstephen80 at gmail.com  Fri Apr 20 20:41:03 2012
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 20 Apr 2012 18:41:03 +0200
Subject: [Freeswitch-users] annexb=yes
In-Reply-To: <CAAB3jWvK-Kh2Uj4kFt7TOZ92y9N1a9quV_-Az915Q_MR3exQ4g@mail.gmail.com>
References: <CAO4LeU0acV8VcN2SwrcogX6=NyFA_4-keraR6agZ8Kvsf-usmQ@mail.gmail.com>
	<CAKzWOxX+x3A79S8vdgQkCRip=BM6PyUmEc02J_=KPxs2-APZkQ@mail.gmail.com>
	<CAO4LeU2a2o=PztcG_HCm2RBbi4woEB5OnKVs5eN=cei_4PcmGw@mail.gmail.com>
	<CAAB3jWvK-Kh2Uj4kFt7TOZ92y9N1a9quV_-Az915Q_MR3exQ4g@mail.gmail.com>
Message-ID: <CAO4LeU0rAJ6SsGsignUsS8ALciiQQODP_vmb55g0ANFYxi_2hw@mail.gmail.com>

Thank you! It works fine!

On Fri, Apr 20, 2012 at 6:13 PM, DJB International <djbinter at gmail.com>wrote:

> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=yes"/>
>
>
>
> On Fri, Apr 20, 2012 at 9:10 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> The call is coming from different providers and I do a bridge to
>> originate the B leg.
>> I want to force in this outgoing leg the annexb=yes: it's possible to
>> force it int outgoing call leg?
>>
>> On Fri, Apr 20, 2012 at 5:46 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>  What device is originating the calls?
>>> -MC
>>>
>>> On Fri, Apr 20, 2012 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> In my Freeswitch, the originated outgoing calls have:
>>>>
>>>> Media Attribute (a): fmtp:18 annexb=no
>>>>
>>>> How can I change it in annexb=yes ?
>>>>
>>>> Stephen
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From wesleyakio at tuntscorp.com  Fri Apr 20 21:10:56 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Fri, 20 Apr 2012 14:10:56 -0300
Subject: [Freeswitch-users] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>
References: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>
Message-ID: <CAGTsDSumzX8bhnZxYiCyvpARhh24Ko+6EbPuz6A7hz741Prc9g@mail.gmail.com>

Hello Giovanni,

I'm off to buy one of those and will let you know how it goes...

In a side note, how reliable so far?

[ ]s,

Wesley Akio
TuntsCorp.com


On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Hello FreeSWITCHers,
>
> the new version of GSMopen is basically usable, and the docs at the
> wiki almost up to date (barring the details of the config file, you
> don't need them now :) ).
>
> Good news is that GSMopen now supports Huawei E1550 and compatible
> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>
> Thanks to dongle's pcm serial streaming via USB, no more need for
> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
> digital.
>
> It's reasonable to expect to be able to support many dozen concurrent
> calls per machine (dongles are USB 2.0 compliant).
>
> Works well right now on Linux and Windows, and would probably works on
> *BSD and OSX with minor tweaks.
>
> Code will be polished, refactured, rewritten, etc. But the thing is there.
>
> So, please update your source code to latest git, have a look at the
> wiki page, build mod_gsmopen, connect a dongle to your machine, and
> begin catching bugs and documentation inaccuracies. And asking for
> features for your use cases.
>
> Thanks to you all in advance, it's a pleasure and an honor to be part
> of this community.
>
> http://wiki.freeswitch.org/wiki/Gsmopen
>
> -giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From kyle.king at quentustech.com  Fri Apr 20 20:17:27 2012
From: kyle.king at quentustech.com (Kyle King)
Date: Fri, 20 Apr 2012 09:17:27 -0700
Subject: [Freeswitch-users] FreeSWITCH mod_fail2ban
Message-ID: <4F918C17.4030909@quentustech.com>

Hello everyone, I've written a FreeSWITCH module for the application
Fail2ban. It currently logs all inbound registration attempts, and
failed registrations to a log file
"/usr/local/freeswitch/log/fail2ban.log". The advantage to using this module
for fail2ban rather than the standard FS log is that you don't have to
keep your server log level as verbose.

If anyone is willing to look it over and give me feedback please do so
at https://github.com/kyconquers/freeswitch-mod_fail2ban

-- 
Kyle A. King
Quentus Technologies, INC
Cell: 703-635-9512
Work: 253-218-6030
Fax:  206-462-1861
Email: kyle.king at quentustech.com



From msc at freeswitch.org  Fri Apr 20 21:25:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 20 Apr 2012 10:25:34 -0700
Subject: [Freeswitch-users] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>
References: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>
Message-ID: <CAKzWOxUivDOL7QUqVjmNydgackb0fn0Q6fkph=J2_6dg8wLpRg@mail.gmail.com>

Great news! FYI, I've got you scheduled for May 9th to come into the FS
community conference call to discuss this subject. For those wanting to
talk live to Giovanni about this subject be sure to order your E1550
dongles right away so that you can be a part of the discussion.

-MC

On Fri, Apr 20, 2012 at 8:58 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Hello FreeSWITCHers,
>
> the new version of GSMopen is basically usable, and the docs at the
> wiki almost up to date (barring the details of the config file, you
> don't need them now :) ).
>
> Good news is that GSMopen now supports Huawei E1550 and compatible
> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>
> Thanks to dongle's pcm serial streaming via USB, no more need for
> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
> digital.
>
> It's reasonable to expect to be able to support many dozen concurrent
> calls per machine (dongles are USB 2.0 compliant).
>
> Works well right now on Linux and Windows, and would probably works on
> *BSD and OSX with minor tweaks.
>
> Code will be polished, refactured, rewritten, etc. But the thing is there.
>
> So, please update your source code to latest git, have a look at the
> wiki page, build mod_gsmopen, connect a dongle to your machine, and
> begin catching bugs and documentation inaccuracies. And asking for
> features for your use cases.
>
> Thanks to you all in advance, it's a pleasure and an honor to be part
> of this community.
>
> http://wiki.freeswitch.org/wiki/Gsmopen
>
> -giovanni
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gmaruzz at gmail.com  Fri Apr 20 21:35:59 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 20 Apr 2012 19:35:59 +0200
Subject: [Freeswitch-users] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CAGTsDSumzX8bhnZxYiCyvpARhh24Ko+6EbPuz6A7hz741Prc9g@mail.gmail.com>
References: <CALXCt0pg1_wVhHO=CLW9U7TACesmHuTCHLrCmTdK8quEez4ubg@mail.gmail.com>
	<CAGTsDSumzX8bhnZxYiCyvpARhh24Ko+6EbPuz6A7hz741Prc9g@mail.gmail.com>
Message-ID: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>

On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
> Hello Giovanni,
>
> I'm off to buy one of those and will let you know how it goes...
>
> In a side note, how reliable so far?

Hardware is very reliable and heavy duty.
Code will become better :)


>
> [ ]s,
>
> Wesley Akio
> TuntsCorp.com
>
>
> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> Hello FreeSWITCHers,
>>
>> the new version of GSMopen is basically usable, and the docs at the
>> wiki almost up to date (barring the details of the config file, you
>> don't need them now :) ).
>>
>> Good news is that GSMopen now supports Huawei E1550 and compatible
>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>>
>> Thanks to dongle's pcm serial streaming via USB, no more need for
>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
>> digital.
>>
>> It's reasonable to expect to be able to support many dozen concurrent
>> calls per machine (dongles are USB 2.0 compliant).
>>
>> Works well right now on Linux and Windows, and would probably works on
>> *BSD and OSX with minor tweaks.
>>
>> Code will be polished, refactured, rewritten, etc. But the thing is there.
>>
>> So, please update your source code to latest git, have a look at the
>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
>> begin catching bugs and documentation inaccuracies. And asking for
>> features for your use cases.
>>
>> Thanks to you all in advance, it's a pleasure and an honor to be part
>> of this community.
>>
>> http://wiki.freeswitch.org/wiki/Gsmopen
>>
>> -giovanni
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From krice at freeswitch.org  Fri Apr 20 21:37:53 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Apr 2012 12:37:53 -0500
Subject: [Freeswitch-users] [Freeswitch-dev]  mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
Message-ID: <CBB70921.31553%krice@freeswitch.org>

Hey Giovanni is there a good usb dongle that's widely available on somewhere
like Amazon.com that you can recommend?

K


On 4/20/12 12:35 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
>> Hello Giovanni,
>> 
>> I'm off to buy one of those and will let you know how it goes...
>> 
>> In a side note, how reliable so far?
> 
> Hardware is very reliable and heavy duty.
> Code will become better :)
> 
> 
>> 
>> [ ]s,
>> 
>> Wesley Akio
>> TuntsCorp.com
>> 
>> 
>> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
>> <gmaruzz at gmail.com>wrote:
>> 
>>> Hello FreeSWITCHers,
>>> 
>>> the new version of GSMopen is basically usable, and the docs at the
>>> wiki almost up to date (barring the details of the config file, you
>>> don't need them now :) ).
>>> 
>>> Good news is that GSMopen now supports Huawei E1550 and compatible
>>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>>> 
>>> Thanks to dongle's pcm serial streaming via USB, no more need for
>>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
>>> digital.
>>> 
>>> It's reasonable to expect to be able to support many dozen concurrent
>>> calls per machine (dongles are USB 2.0 compliant).
>>> 
>>> Works well right now on Linux and Windows, and would probably works on
>>> *BSD and OSX with minor tweaks.
>>> 
>>> Code will be polished, refactured, rewritten, etc. But the thing is there.
>>> 
>>> So, please update your source code to latest git, have a look at the
>>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
>>> begin catching bugs and documentation inaccuracies. And asking for
>>> features for your use cases.
>>> 
>>> Thanks to you all in advance, it's a pleasure and an honor to be part
>>> of this community.
>>> 
>>> http://wiki.freeswitch.org/wiki/Gsmopen
>>> 
>>> -giovanni
>>> 
>>> 
>>> --
>>> Sincerely,
>>> 
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
> 




From tlanghals at gmail.com  Fri Apr 20 21:38:34 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Fri, 20 Apr 2012 12:38:34 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <CAKbxfG_H_=maHQ67s7EEY5-CsFf4QJJryFmDJn1vTHdkXd2cCw@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
	<CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
	<4F918928.5080608@communicatefreely.net>
	<CAKbxfG_H_=maHQ67s7EEY5-CsFf4QJJryFmDJn1vTHdkXd2cCw@mail.gmail.com>
Message-ID: <CAJ_WQfHsZUc8arsOUYAgda2_OO9iKxfm80EvEeSivEoN2N5rzg@mail.gmail.com>

I'm using the server IP address for the domain in all configuration at this
point and the BLF does work correctly when I set the key value to
queue+park_slot1 but does not work when I set the value to park_slot1 so
doesn't seem like it would be a domain issue.  I'm currently not using ODBC
for presence or the force domain option, is ODBC required?  All
configuration in the Aastra uses the IP address

I'll try changing domain=* and see if it makes a difference.  I think my
regex should be fine <exten regex="park_slot\d+" proto="queue"/> would
match park_slot1

Do you mind copying a piece of your presence_map.conf.xml config for
comparison?

Thanks for the assistance.

On Fri, Apr 20, 2012 at 11:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You might want to stick with the domain=* unless you know the hosts in
> your sip packet match the $${domain} for sure.
> it's "queue" for fifo and you need to make sure the regex matches any
> extensions in question that you are subscribing to.
>
> On Fri, Apr 20, 2012 at 11:04 AM, Tim St. Pierre
> <fs-list at communicatefreely.net> wrote:
> > I'm also using the Aastra phones, and I had to use the presence map to
> > make things work.
> >
> > proto="queue" is correct.
> > My regex is a bit different, as my parking slots have slightly different
> > naming conventions, but it works.
> >
> > You may want to look at the subscriptions to make sure the domains
> > match.  Are you using ODBC for sofia?  If so, you can look in the
> > sip_subscriptions table and see the domain that was subscribed to.  That
> > domain has to be the same as the domain you use in the presence map.
> >
> > Also in your sofia profile, there is an option to force subscription
> > domain.  Are you using that?  You don't have to if the domains match.
> >
> > With the Aastra phones, the domain used in the subscribe and also in
> > invites is whatever you put in sip proxy and sip registrar (still
> > haven't figured out which is which).  Are you using your SIP domain, or
> > an IP address?  If you are using an IP address, make the proxy and
> > registrar fields the SIP domain, and put the IP address and port in the
> > outbound proxy section.
> >
> > Hope that's helpful.
> >
> > -Tim
> >
> > Travis Langhals wrote:
> >> I added <param name="presence-proto-lookup" value="true"/> to my sip
> >> interface config and modified presence_map.conf.xml as follows but
> >> still no luck.  I tried proto="queue", proto="queue+", proto="fifo"
> >> and restarted freeswitch between each test.  My fifos are named
> >> park_slot1, park_slot2, etc.
> >>
> >> <configuration name="presence_map.conf" description="PRESENCE MAP">
> >>   <domains>
> >>     <domain name="$${domain}">
> >>       <exten regex="park_slot\d+" proto="queue"/>
> >>     </domain>
> >>   </domains>
> >> </configuration>
> >>
> >>
> >> On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
> wrote:
> >>
> >>     look in tree for the installed presence_map.conf.xml
> >>
> >>
> >>     its a series of ext range regex vs protos
> >>
> >>
> >>     you must also set the sofia profile param presence-proto-lookup to
> >>     true
> >>
> >>
> >>
> >>     On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals
> >>     <tlanghals at gmail.com <mailto:tlanghals at gmail.com>> wrote:
> >>     > I've setup a fifo park queue based off the example at
> >>     > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying
> >>     to use the
> >>     > Aastra BLF/Xfer key to interface with it.  The parking fifo
> >>     queue works
> >>     > great but I have to use two keys, one for BLF to subscribe to
> >>     > queue+<fifo_name> and another for speed dial/tranfer to
> <fifo_name>.
> >>     >
> >>     > Is there any way to remove the need to have the queue+ prefix
> >>     needed to
> >>     > subscribe to the fifo presence so that I can just use a single
> >>     BLF/Xfer key
> >>     > point to <fifo_name>?
> >>     >
> >>     > I found this thread that mentions using the map proto feature to
> >>     eliminate
> >>     > the need for the prefix but cannot find anything information
> >>     relating to
> >>     > such a feature.
> >>     >
> >>
> http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
> >>     >
> >>     >
> >>
> _________________________________________________________________________
> >>     > Professional FreeSWITCH Consulting Services:
> >>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>     > http://www.freeswitchsolutions.com
> >>     >
> >>     > 
> >>     > 
> >>     >
> >>     > Official FreeSWITCH Sites
> >>     > http://www.freeswitch.org
> >>     > http://wiki.freeswitch.org
> >>     > http://www.cluecon.com
> >>     >
> >>     > FreeSWITCH-users mailing list
> >>     > FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     >
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     > http://www.freeswitch.org
> >>     >
> >>
> >>
> >>
> >>     --
> >>     Anthony Minessale II
> >>
> >>     FreeSWITCH http://www.freeswitch.org/
> >>     ClueCon http://www.cluecon.com/
> >>     Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >>     AIM: anthm
> >>     MSN:anthony_minessale at hotmail.com
> >>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
> >>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> >>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >>     FreeSWITCH Developer Conference
> >>     sip:888 at conference.freeswitch.org
> >>     <mailto:sip%3A888 at conference.freeswitch.org>
> >>     googletalk:conf+888 at conference.freeswitch.org
> >>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>     pstn:+19193869900 <tel:%2B19193869900>
> >>
> >>
> _________________________________________________________________________
> >>     Professional FreeSWITCH Consulting Services:
> >>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>     http://www.freeswitchsolutions.com
> >>
> >>     
> >>     
> >>
> >>     Official FreeSWITCH Sites
> >>     http://www.freeswitch.org
> >>     http://wiki.freeswitch.org
> >>     http://www.cluecon.com
> >>
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>     http://www.freeswitch.org
> >>
> >>
> >> ------------------------------------------------------------------------
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From krice at freeswitch.org  Fri Apr 20 21:40:08 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 20 Apr 2012 12:40:08 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH mod_fail2ban
In-Reply-To: <4F918C17.4030909@quentustech.com>
Message-ID: <CBB709A8.31555%krice@freeswitch.org>

Hey Kyle Love this project... I would make a few suggestions... Possibly
could help with a couple of patches can you find me on IRC as SWK?

K


On 4/20/12 11:17 AM, "Kyle King" <kyle.king at quentustech.com> wrote:

> Hello everyone, I've written a FreeSWITCH module for the application
> Fail2ban. It currently logs all inbound registration attempts, and
> failed registrations to a log file
> "/usr/local/freeswitch/log/fail2ban.log". The advantage to using this module
> for fail2ban rather than the standard FS log is that you don't have to
> keep your server log level as verbose.
> 
> If anyone is willing to look it over and give me feedback please do so
> at https://github.com/kyconquers/freeswitch-mod_fail2ban




From gmaruzz at gmail.com  Fri Apr 20 21:43:34 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 20 Apr 2012 19:43:34 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CBB70921.31553%krice@freeswitch.org>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
Message-ID: <CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>

Huawei E1550 or compatibles.
I'll write a list of compatible dongles maybe tomorrow.
-g


On 4/20/12, Ken Rice <krice at freeswitch.org> wrote:
> Hey Giovanni is there a good usb dongle that's widely available on somewhere
> like Amazon.com that you can recommend?
>
> K
>
>
> On 4/20/12 12:35 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
>>> Hello Giovanni,
>>>
>>> I'm off to buy one of those and will let you know how it goes...
>>>
>>> In a side note, how reliable so far?
>>
>> Hardware is very reliable and heavy duty.
>> Code will become better :)
>>
>>
>>>
>>> [ ]s,
>>>
>>> Wesley Akio
>>> TuntsCorp.com
>>>
>>>
>>> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
>>> <gmaruzz at gmail.com>wrote:
>>>
>>>> Hello FreeSWITCHers,
>>>>
>>>> the new version of GSMopen is basically usable, and the docs at the
>>>> wiki almost up to date (barring the details of the config file, you
>>>> don't need them now :) ).
>>>>
>>>> Good news is that GSMopen now supports Huawei E1550 and compatible
>>>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>>>>
>>>> Thanks to dongle's pcm serial streaming via USB, no more need for
>>>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
>>>> digital.
>>>>
>>>> It's reasonable to expect to be able to support many dozen concurrent
>>>> calls per machine (dongles are USB 2.0 compliant).
>>>>
>>>> Works well right now on Linux and Windows, and would probably works on
>>>> *BSD and OSX with minor tweaks.
>>>>
>>>> Code will be polished, refactured, rewritten, etc. But the thing is
>>>> there.
>>>>
>>>> So, please update your source code to latest git, have a look at the
>>>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
>>>> begin catching bugs and documentation inaccuracies. And asking for
>>>> features for your use cases.
>>>>
>>>> Thanks to you all in advance, it's a pleasure and an honor to be part
>>>> of this community.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Gsmopen
>>>>
>>>> -giovanni
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From wesleyakio at tuntscorp.com  Fri Apr 20 22:32:36 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Fri, 20 Apr 2012 15:32:36 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
Message-ID: <CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>

Hi Giovanni,

I've hit the local stores and got ZTEs, NOVIASs, SIM!s, and all sorts off
stuf, but no Huaweys...

That compatibility list would come in handy!

Best,

Wesley Akio
TuntsCorp.com


On Fri, Apr 20, 2012 at 2:43 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Huawei E1550 or compatibles.
> I'll write a list of compatible dongles maybe tomorrow.
> -g
>
>
> On 4/20/12, Ken Rice <krice at freeswitch.org> wrote:
> > Hey Giovanni is there a good usb dongle that's widely available on
> somewhere
> > like Amazon.com that you can recommend?
> >
> > K
> >
> >
> > On 4/20/12 12:35 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
> >
> >> On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
> >>> Hello Giovanni,
> >>>
> >>> I'm off to buy one of those and will let you know how it goes...
> >>>
> >>> In a side note, how reliable so far?
> >>
> >> Hardware is very reliable and heavy duty.
> >> Code will become better :)
> >>
> >>
> >>>
> >>> [ ]s,
> >>>
> >>> Wesley Akio
> >>> TuntsCorp.com
> >>>
> >>>
> >>> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
> >>> <gmaruzz at gmail.com>wrote:
> >>>
> >>>> Hello FreeSWITCHers,
> >>>>
> >>>> the new version of GSMopen is basically usable, and the docs at the
> >>>> wiki almost up to date (barring the details of the config file, you
> >>>> don't need them now :) ).
> >>>>
> >>>> Good news is that GSMopen now supports Huawei E1550 and compatible
> >>>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
> >>>>
> >>>> Thanks to dongle's pcm serial streaming via USB, no more need for
> >>>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
> >>>> digital.
> >>>>
> >>>> It's reasonable to expect to be able to support many dozen concurrent
> >>>> calls per machine (dongles are USB 2.0 compliant).
> >>>>
> >>>> Works well right now on Linux and Windows, and would probably works on
> >>>> *BSD and OSX with minor tweaks.
> >>>>
> >>>> Code will be polished, refactured, rewritten, etc. But the thing is
> >>>> there.
> >>>>
> >>>> So, please update your source code to latest git, have a look at the
> >>>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
> >>>> begin catching bugs and documentation inaccuracies. And asking for
> >>>> features for your use cases.
> >>>>
> >>>> Thanks to you all in advance, it's a pleasure and an honor to be part
> >>>> of this community.
> >>>>
> >>>> http://wiki.freeswitch.org/wiki/Gsmopen
> >>>>
> >>>> -giovanni
> >>>>
> >>>>
> >>>> --
> >>>> Sincerely,
> >>>>
> >>>> Giovanni Maruzzelli
> >>>> Cell : +39-347-2665618
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> 
> >>>> 
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://wiki.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From acrow at integrafin.co.uk  Fri Apr 20 23:48:19 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Fri, 20 Apr 2012 20:48:19 +0100
Subject: [Freeswitch-users] Originate call into conference - no early
 media
In-Reply-To: <4F911932.7030002@integrafin.co.uk>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F911932.7030002@integrafin.co.uk>
Message-ID: <4F91BD83.7000405@integrafin.co.uk>

On 20/04/12 09:07, Alex Crow wrote:
> On 20/04/12 01:37, Anthony Minessale wrote:
>> You only get it when you actually are being sent early media.
>> Probably the ones that dont have any sound are only sending 180 so
>> there is nothing you can do about it.
>>
> Anthony,
>
> Thanks - would this only be "unfixable" for internal phones, as I'm sure
> I could change at least my other freeswitch boxes I am gatewaying to to
> send early media?
>
> Cheers
>
> Alex
>

Anthony,

I've managed to fix this for calling out to my FS gateways by correcting 
a mistake on their Local_Extensions (ring_ready=${uk-ring}, should have 
been ringback=${uk-ring})

However I am suprised there is no way to fake ringback for a conference 
dialout to endpoints that only reply with 180. Is there no workaround 
for this? I've tried using the conference dial api command and I get the 
same result.

Cheers

Alex


From vipkilla at gmail.com  Sat Apr 21 00:08:19 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 20 Apr 2012 16:08:19 -0400
Subject: [Freeswitch-users] Memory usage of mod_local_stream
Message-ID: <CAC-LwPM9gRmR11++M5jhGJ3Naao7cQ7o=JbWOheMz_m2eU7_AQ@mail.gmail.com>

I'm wondering how many local_stream FS can handle with the following parameters:
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="rate" value="8000"/>
<param name="timer-name" value="soft"/>

Is there a way to calculate or know how much memory each stream will
use with these settings?


From rmorin at blie-ent.com  Sat Apr 21 00:53:45 2012
From: rmorin at blie-ent.com (Rob Morin)
Date: Fri, 20 Apr 2012 16:53:45 -0400
Subject: [Freeswitch-users] Dropped calls
Message-ID: <11e701cd1f37$ae02e4e0$0a08aea0$@blie-ent.com>

I recently (4/17/2012) did a "make current" on my Freeswitch PBX. (Last time
I had done it was probably in January.) No particular reason for the update
- just timely.

 

After the update, we're now seeing calls drop consistently at the ~1:47
mark. Obviously, this wasn't the case beforehand.

 

I can provide a pcap file and Freeswitch log of a call demonstrating the
issue; I'm just not sure what the best method to do so is.

 

The FS server is sending a BYE, the REASON is 'SIP; cause=408; text="Session
timeout"'.

 

On my end, I have a Grandstream 502 and I can see the keep-alives are
working. But now it is getting disconnected while it wasn't before the
update.

 

Any ideas?

Thank you,

Rob

 

 

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From ga at steadfasttelecom.com  Sat Apr 21 01:02:47 2012
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Fri, 20 Apr 2012 17:02:47 -0400
Subject: [Freeswitch-users] Dropped calls
In-Reply-To: <11e701cd1f37$ae02e4e0$0a08aea0$@blie-ent.com>
References: <11e701cd1f37$ae02e4e0$0a08aea0$@blie-ent.com>
Message-ID: <CAKD7_2RUyAscTys_h=z=PbLiw3zAoP0cQY6psugjC06d4MWDLQ@mail.gmail.com>

Hey Rob,

I think you need to set your grandstreams session timers to UAC. Its on the
page where you setup the extension.

Hope that works
Gill

On Fri, Apr 20, 2012 at 4:53 PM, Rob Morin <rmorin at blie-ent.com> wrote:

> I recently (4/17/2012) did a ?make current? on my Freeswitch PBX. (Last
> time I had done it was probably in January.) No particular reason for the
> update ? just timely.****
>
> ** **
>
> After the update, we?re now seeing calls drop consistently at the ~1:47
> mark. Obviously, this wasn?t the case beforehand.****
>
> ** **
>
> I can provide a pcap file and Freeswitch log of a call demonstrating the
> issue; I?m just not sure what the best method to do so is.****
>
> ** **
>
> The FS server is sending a BYE, the REASON is ?SIP; cause=408;
> text=?Session timeout??.****
>
> ** **
>
> On my end, I have a Grandstream 502 and I can see the keep-alives are
> working. But now it is getting disconnected while it wasn?t before the
> update.****
>
> ** **
>
> Any ideas?
>
> Thank you,****
>
> Rob****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Gilad Abada

SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!

V: 212.589.1001
F: 212.589.1011


For 35 years, Steadfast Telecommunications has been providing
state-of-the-art communications technology to businesses and government
agencies - large and small. Steadfast Telecommunications tailors Unified
Communications and Voice-Over IP Solutions to single-site offices or
multi-site and worldwide enterprises.   Make your virtual office a
reality.  Enjoy the freedom to travel while remaining connected to your
office.
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From anita.hall at simmortel.com  Sat Apr 21 01:05:46 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Sat, 21 Apr 2012 02:35:46 +0530
Subject: [Freeswitch-users] unimrcp nuance grammar failed to load
Message-ID: <CAGKPtBuGAT5HOGpj9xu4OkH=VpHd1Pg=EE5K0VDPKLcyjkx0dg@mail.gmail.com>

Hi

I am getting the following error when using detect_speech from dialplan
with unimrcp, nuance speech server and nuance recognizer. I understand that
it is the job of the unimrcp client (ie mod_unimrcp) to convert the grammar
from JSGF to MRCP. Why do I not see any grammar text in the SIP trace ?
What could be wrong here ?

2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:3433 (ASR-0) grammar
failed to load, status code = 407
2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:1541 (ASR-0) PROCESSING
==> ERROR

Complete logs are here
http://pastebin.freeswitch.org/18900


The SIP trace : (10.60.20.25 is FreeSWITCH and 10.60.20.47 is Nuance Speech
Server)
10.60.20.25 -> 10.60.20.47  SIP/SDP Request: INVITE sip:10.60.20.47:5060,
with session description
10.60.20.47 -> 10.60.20.25  SIP/SDP Status: 200 OK, with session description
10.60.20.47 -> 10.60.20.25  SIP Request: BYE sip:10.60.20.25:5090
10.60.20.25 -> 10.60.20.47  SIP Status: 481 Call Does Not Exist



My dialplan is
                        <action application="answer"/>
                        <action application="detect_speech"
data="unimrcp:nuance5-mrcp2 yesno yesno.gram"/>
                        <action application="detect_speech" data="grammar
yesno"/>
                        <action application="detect_speech" data="grammaron
yesno"/>
                        <action application="sleep" data="500"/>

My grammar is yesno.gram kept in /usr/local/freeswitch/grammar

#JSGF V1.0;
grammar example;
<yes> = [ yes ];
<no> = [ no ];
public <results> = [ <yes> | <no> ];


# cat conf/autoload_configs/unimrcp.conf.xml
is here
http://pastebin.freeswitch.org/18899

# cat conf/mrcp_profiles/nuance-5.0-mrcp-v2.xml
is here
http://pastebin.freeswitch.org/18898







regards,
Anita
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From bdfoster at endigotech.com  Sat Apr 21 02:35:01 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 20 Apr 2012 18:35:01 -0400
Subject: [Freeswitch-users] Best practice for paging in an environment
 with multiple-registrations enabled on Sofia profile
In-Reply-To: <4F91899D.9020104@communicatefreely.net>
References: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
	<4F91899D.9020104@communicatefreely.net>
Message-ID: <CAKV-jFyYHGdtpeKb8XWsUeidqAHAsm4EOR-WNjRb0SGy+DRoSg@mail.gmail.com>

Multicast isn't really the best solution, because there may be others that
are not on the LAN.
On Apr 20, 2012 12:07 PM, "Tim St. Pierre" <fs-list at communicatefreely.net>
wrote:

> Not to change the subject, but is multi-cast an option for you?  That
> solves a lot of problems, and Freeswitch is capable of generating the
> multi-cast stream.
>
> -Tim
>
> Brian Foster wrote:
> >
> > As the subject suggests, what is the best way to do paging with
> > multiple-registrations enabled? In my lab I have two phones, one is
> > registered to 2546 and 2686, the other is registered to 2546 and 2738
> > (so both phones are registered to 2546 and another extension). I have
> > something set up so that FS sends out a page like so:
> >
> > originate {sip_auto_answer=true}user/2546 at domain.tld
> > &playback(alertmessage.wav)
> >
> > However as you might expect (it does make sense) the first phone to
> > win the lottery and pick up the phone first wins the call and auto
> > answers the call. How do I properly get every extension to playback
> > the message at the same time?
> >
> > -BDF
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From tlanghals at gmail.com  Sat Apr 21 04:33:02 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Fri, 20 Apr 2012 19:33:02 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <CAJ_WQfHsZUc8arsOUYAgda2_OO9iKxfm80EvEeSivEoN2N5rzg@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
	<CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
	<4F918928.5080608@communicatefreely.net>
	<CAKbxfG_H_=maHQ67s7EEY5-CsFf4QJJryFmDJn1vTHdkXd2cCw@mail.gmail.com>
	<CAJ_WQfHsZUc8arsOUYAgda2_OO9iKxfm80EvEeSivEoN2N5rzg@mail.gmail.com>
Message-ID: <CAJ_WQfGEJQ7QbpOMYymu0WtvndRbUP8qxMgCbAgxRVLrDSpCRA@mail.gmail.com>

Now I'm really confused, it's all of a sudden working with my original
config posted.  But further I tried to break it by changing the extension
map regex and restarting everything yet it still works????  Is the non odbc
presence info saved somewhere?

On Fri, Apr 20, 2012 at 12:38 PM, Travis Langhals <tlanghals at gmail.com>wrote:

> I'm using the server IP address for the domain in all configuration at
> this point and the BLF does work correctly when I set the key value to
> queue+park_slot1 but does not work when I set the value to park_slot1 so
> doesn't seem like it would be a domain issue.  I'm currently not using ODBC
> for presence or the force domain option, is ODBC required?  All
> configuration in the Aastra uses the IP address
>
> I'll try changing domain=* and see if it makes a difference.  I think my
> regex should be fine <exten regex="park_slot\d+" proto="queue"/> would
> match park_slot1
>
> Do you mind copying a piece of your presence_map.conf.xml config for
> comparison?
>
> Thanks for the assistance.
>
>
> On Fri, Apr 20, 2012 at 11:20 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> You might want to stick with the domain=* unless you know the hosts in
>> your sip packet match the $${domain} for sure.
>> it's "queue" for fifo and you need to make sure the regex matches any
>> extensions in question that you are subscribing to.
>>
>> On Fri, Apr 20, 2012 at 11:04 AM, Tim St. Pierre
>> <fs-list at communicatefreely.net> wrote:
>> > I'm also using the Aastra phones, and I had to use the presence map to
>> > make things work.
>> >
>> > proto="queue" is correct.
>> > My regex is a bit different, as my parking slots have slightly different
>> > naming conventions, but it works.
>> >
>> > You may want to look at the subscriptions to make sure the domains
>> > match.  Are you using ODBC for sofia?  If so, you can look in the
>> > sip_subscriptions table and see the domain that was subscribed to.  That
>> > domain has to be the same as the domain you use in the presence map.
>> >
>> > Also in your sofia profile, there is an option to force subscription
>> > domain.  Are you using that?  You don't have to if the domains match.
>> >
>> > With the Aastra phones, the domain used in the subscribe and also in
>> > invites is whatever you put in sip proxy and sip registrar (still
>> > haven't figured out which is which).  Are you using your SIP domain, or
>> > an IP address?  If you are using an IP address, make the proxy and
>> > registrar fields the SIP domain, and put the IP address and port in the
>> > outbound proxy section.
>> >
>> > Hope that's helpful.
>> >
>> > -Tim
>> >
>> > Travis Langhals wrote:
>> >> I added <param name="presence-proto-lookup" value="true"/> to my sip
>> >> interface config and modified presence_map.conf.xml as follows but
>> >> still no luck.  I tried proto="queue", proto="queue+", proto="fifo"
>> >> and restarted freeswitch between each test.  My fifos are named
>> >> park_slot1, park_slot2, etc.
>> >>
>> >> <configuration name="presence_map.conf" description="PRESENCE MAP">
>> >>   <domains>
>> >>     <domain name="$${domain}">
>> >>       <exten regex="park_slot\d+" proto="queue"/>
>> >>     </domain>
>> >>   </domains>
>> >> </configuration>
>> >>
>> >>
>> >> On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
>> wrote:
>> >>
>> >>     look in tree for the installed presence_map.conf.xml
>> >>
>> >>
>> >>     its a series of ext range regex vs protos
>> >>
>> >>
>> >>     you must also set the sofia profile param presence-proto-lookup to
>> >>     true
>> >>
>> >>
>> >>
>> >>     On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals
>> >>     <tlanghals at gmail.com <mailto:tlanghals at gmail.com>> wrote:
>> >>     > I've setup a fifo park queue based off the example at
>> >>     > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying
>> >>     to use the
>> >>     > Aastra BLF/Xfer key to interface with it.  The parking fifo
>> >>     queue works
>> >>     > great but I have to use two keys, one for BLF to subscribe to
>> >>     > queue+<fifo_name> and another for speed dial/tranfer to
>> <fifo_name>.
>> >>     >
>> >>     > Is there any way to remove the need to have the queue+ prefix
>> >>     needed to
>> >>     > subscribe to the fifo presence so that I can just use a single
>> >>     BLF/Xfer key
>> >>     > point to <fifo_name>?
>> >>     >
>> >>     > I found this thread that mentions using the map proto feature to
>> >>     eliminate
>> >>     > the need for the prefix but cannot find anything information
>> >>     relating to
>> >>     > such a feature.
>> >>     >
>> >>
>> http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>> >>     >
>> >>     >
>> >>
>> _________________________________________________________________________
>> >>     > Professional FreeSWITCH Consulting Services:
>> >>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >>     > http://www.freeswitchsolutions.com
>> >>     >
>> >>     > 
>> >>     > 
>> >>     >
>> >>     > Official FreeSWITCH Sites
>> >>     > http://www.freeswitch.org
>> >>     > http://wiki.freeswitch.org
>> >>     > http://www.cluecon.com
>> >>     >
>> >>     > FreeSWITCH-users mailing list
>> >>     > FreeSWITCH-users at lists.freeswitch.org
>> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >
>> >>     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>     > http://www.freeswitch.org
>> >>     >
>> >>
>> >>
>> >>
>> >>     --
>> >>     Anthony Minessale II
>> >>
>> >>     FreeSWITCH http://www.freeswitch.org/
>> >>     ClueCon http://www.cluecon.com/
>> >>     Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >>     AIM: anthm
>> >>     MSN:anthony_minessale at hotmail.com
>> >>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> >>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> >>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>> >>
>> >>     FreeSWITCH Developer Conference
>> >>     sip:888 at conference.freeswitch.org
>> >>     <mailto:sip%3A888 at conference.freeswitch.org>
>> >>     googletalk:conf+888 at conference.freeswitch.org
>> >>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>     pstn:+19193869900 <tel:%2B19193869900>
>> >>
>> >>
>> _________________________________________________________________________
>> >>     Professional FreeSWITCH Consulting Services:
>> >>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >>     http://www.freeswitchsolutions.com
>> >>
>> >>     
>> >>     
>> >>
>> >>     Official FreeSWITCH Sites
>> >>     http://www.freeswitch.org
>> >>     http://wiki.freeswitch.org
>> >>     http://www.cluecon.com
>> >>
>> >>     FreeSWITCH-users mailing list
>> >>     FreeSWITCH-users at lists.freeswitch.org
>> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>     http://www.freeswitch.org
>> >>
>> >>
>> >>
>> ------------------------------------------------------------------------
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From fieldpeak at gmail.com  Sat Apr 21 04:35:49 2012
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 21 Apr 2012 08:35:49 +0800
Subject: [Freeswitch-users] recommend free/open source sip client for
	iPhone
In-Reply-To: <CAOTmbMySwMsyHYh3768pRXwEMJvKLP5QCfE9+jfvYUWU20iy9A@mail.gmail.com>
References: <CAOTmbMyUSdqEH2NFz02r228pEFNt0ZO4prSoscvY7YnD-9=4sA@mail.gmail.com>
	<CAO4LeU0R45q9zfTgrhoLADxX74h6ynErzRMvQLaigPekjc3s6g@mail.gmail.com>
	<614671334828621@web9e.yandex.ru>
	<CAO4LeU3VcnyP11hFF+fRLz_yGGou3WH=AiZEGedLuef0Qi0nsg@mail.gmail.com>
	<72171334833143@web10e.yandex.ru>
	<CAO4LeU3++c-CZjbm0Cq=pd0aEHZ3o9H1RbvHJsjXiHXM7WUcVA@mail.gmail.com>
	<CAOTmbMySwMsyHYh3768pRXwEMJvKLP5QCfE9+jfvYUWU20iy9A@mail.gmail.com>
Message-ID: <CAOTmbMyTAruS=DOUk=cGYD7MuHmisZ5GEwyBQ-B_7=-S1H+-mQ@mail.gmail.com>

Afer tested a lot, Zoiper is really a good one, free and suport background
receiving call.

Appreciate very much for Dorn' recommendation.
? 2012-4-19 ??8:52?"fieldpeak" <fieldpeak at gmail.com>???

> It is really a amazing topic, so many people involved, thank you all :)
> welcome more recommendation...
>
> Regards,
> Charles
>
> 2012/4/19 Stephen Wilde <wstephen80 at gmail.com>
>
>> It require an account to their server.
>> The call goes probally goes via them.
>>
>> On Thu, Apr 19, 2012 at 12:59 PM, Serge S. Yuriev <me at nevian.org> wrote:
>>
>>> IIRC even in this case traffic goes via them, isn't it?
>>> At least it requires account on ther server
>>>
>>> 19.04.2012, 14:05, "Stephen Wilde" <wstephen80 at gmail.com>:
>>>
>>> I'm using Fring on iPhone with my FreeSwitch. In Fring "AddOn" you can
>>> configure a normal SIP account.
>>> Stephen
>>>
>>> On Thu, Apr 19, 2012 at 11:43 AM, Serge S. Yuriev <me at nevian.org> wrote:
>>>
>>>  I'm afraid Fring is compete complete platform which doesn't have direct
>>> SIP connection (all traffic goes via their servers)
>>>
>>> 19.04.2012, 11:54, "Stephen Wilde" <wstephen80 at gmail.com>:
>>>
>>> Fring
>>>
>>> On Tue, Apr 17, 2012 at 3:27 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>> Is there any recommended free/open source sip client for iPhone? Thanks.
>>>
>>>  --
>>> wbr,
>>> Serge
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Regards,
> Charles
>
>
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From tlanghals at gmail.com  Sat Apr 21 04:52:03 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Fri, 20 Apr 2012 19:52:03 -0500
Subject: [Freeswitch-users] How to use alternate presence id for a fifo
	queue?
In-Reply-To: <CAJ_WQfGEJQ7QbpOMYymu0WtvndRbUP8qxMgCbAgxRVLrDSpCRA@mail.gmail.com>
References: <CAJ_WQfFD4eWPhNpsVuKRK97BqGBgZycDbt-Zo57vNNWhg_GA-g@mail.gmail.com>
	<CAKbxfG_ke7EOkV1SyaDgXZ1ZN_4kL4VmjA82uVrmhwqygzZUVw@mail.gmail.com>
	<CAJ_WQfEOy8Dwjc9BZke-zhkreSA3rPsf0APwDB6ruTm3D=g3XA@mail.gmail.com>
	<4F918928.5080608@communicatefreely.net>
	<CAKbxfG_H_=maHQ67s7EEY5-CsFf4QJJryFmDJn1vTHdkXd2cCw@mail.gmail.com>
	<CAJ_WQfHsZUc8arsOUYAgda2_OO9iKxfm80EvEeSivEoN2N5rzg@mail.gmail.com>
	<CAJ_WQfGEJQ7QbpOMYymu0WtvndRbUP8qxMgCbAgxRVLrDSpCRA@mail.gmail.com>
Message-ID: <CAJ_WQfFDN24L1Ky46eb7ETyo5qQ_CXZP06R-pQ4pY3MnLa9reg@mail.gmail.com>

I think I finally got it, the phone was not really resetting, just appeared
to be as it was cached in Firefox and I wasn't paying attention to the
actual phone.  After some additional testing the phone must actually be
reset when changing the presence mapping info to see the changes.

On Fri, Apr 20, 2012 at 7:33 PM, Travis Langhals <tlanghals at gmail.com>wrote:

> Now I'm really confused, it's all of a sudden working with my original
> config posted.  But further I tried to break it by changing the extension
> map regex and restarting everything yet it still works????  Is the non odbc
> presence info saved somewhere?
>
>
> On Fri, Apr 20, 2012 at 12:38 PM, Travis Langhals <tlanghals at gmail.com>wrote:
>
>> I'm using the server IP address for the domain in all configuration at
>> this point and the BLF does work correctly when I set the key value to
>> queue+park_slot1 but does not work when I set the value to park_slot1 so
>> doesn't seem like it would be a domain issue.  I'm currently not using ODBC
>> for presence or the force domain option, is ODBC required?  All
>> configuration in the Aastra uses the IP address
>>
>> I'll try changing domain=* and see if it makes a difference.  I think my
>> regex should be fine <exten regex="park_slot\d+" proto="queue"/> would
>> match park_slot1
>>
>> Do you mind copying a piece of your presence_map.conf.xml config for
>> comparison?
>>
>> Thanks for the assistance.
>>
>>
>> On Fri, Apr 20, 2012 at 11:20 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> You might want to stick with the domain=* unless you know the hosts in
>>> your sip packet match the $${domain} for sure.
>>> it's "queue" for fifo and you need to make sure the regex matches any
>>> extensions in question that you are subscribing to.
>>>
>>> On Fri, Apr 20, 2012 at 11:04 AM, Tim St. Pierre
>>> <fs-list at communicatefreely.net> wrote:
>>> > I'm also using the Aastra phones, and I had to use the presence map to
>>> > make things work.
>>> >
>>> > proto="queue" is correct.
>>> > My regex is a bit different, as my parking slots have slightly
>>> different
>>> > naming conventions, but it works.
>>> >
>>> > You may want to look at the subscriptions to make sure the domains
>>> > match.  Are you using ODBC for sofia?  If so, you can look in the
>>> > sip_subscriptions table and see the domain that was subscribed to.
>>>  That
>>> > domain has to be the same as the domain you use in the presence map.
>>> >
>>> > Also in your sofia profile, there is an option to force subscription
>>> > domain.  Are you using that?  You don't have to if the domains match.
>>> >
>>> > With the Aastra phones, the domain used in the subscribe and also in
>>> > invites is whatever you put in sip proxy and sip registrar (still
>>> > haven't figured out which is which).  Are you using your SIP domain, or
>>> > an IP address?  If you are using an IP address, make the proxy and
>>> > registrar fields the SIP domain, and put the IP address and port in the
>>> > outbound proxy section.
>>> >
>>> > Hope that's helpful.
>>> >
>>> > -Tim
>>> >
>>> > Travis Langhals wrote:
>>> >> I added <param name="presence-proto-lookup" value="true"/> to my sip
>>> >> interface config and modified presence_map.conf.xml as follows but
>>> >> still no luck.  I tried proto="queue", proto="queue+", proto="fifo"
>>> >> and restarted freeswitch between each test.  My fifos are named
>>> >> park_slot1, park_slot2, etc.
>>> >>
>>> >> <configuration name="presence_map.conf" description="PRESENCE MAP">
>>> >>   <domains>
>>> >>     <domain name="$${domain}">
>>> >>       <exten regex="park_slot\d+" proto="queue"/>
>>> >>     </domain>
>>> >>   </domains>
>>> >> </configuration>
>>> >>
>>> >>
>>> >> On Thu, Apr 19, 2012 at 7:24 PM, Anthony Minessale
>>> >> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
>>> wrote:
>>> >>
>>> >>     look in tree for the installed presence_map.conf.xml
>>> >>
>>> >>
>>> >>     its a series of ext range regex vs protos
>>> >>
>>> >>
>>> >>     you must also set the sofia profile param presence-proto-lookup to
>>> >>     true
>>> >>
>>> >>
>>> >>
>>> >>     On Thu, Apr 19, 2012 at 4:00 PM, Travis Langhals
>>> >>     <tlanghals at gmail.com <mailto:tlanghals at gmail.com>> wrote:
>>> >>     > I've setup a fifo park queue based off the example at
>>> >>     > http://wiki.freeswitch.org/wiki/Park_%26_Retrieve and am trying
>>> >>     to use the
>>> >>     > Aastra BLF/Xfer key to interface with it.  The parking fifo
>>> >>     queue works
>>> >>     > great but I have to use two keys, one for BLF to subscribe to
>>> >>     > queue+<fifo_name> and another for speed dial/tranfer to
>>> <fifo_name>.
>>> >>     >
>>> >>     > Is there any way to remove the need to have the queue+ prefix
>>> >>     needed to
>>> >>     > subscribe to the fifo presence so that I can just use a single
>>> >>     BLF/Xfer key
>>> >>     > point to <fifo_name>?
>>> >>     >
>>> >>     > I found this thread that mentions using the map proto feature to
>>> >>     eliminate
>>> >>     > the need for the prefix but cannot find anything information
>>> >>     relating to
>>> >>     > such a feature.
>>> >>     >
>>> >>
>>> http://jira.freeswitch.org/browse/FS-2831?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>>> >>     >
>>> >>     >
>>> >>
>>> _________________________________________________________________________
>>> >>     > Professional FreeSWITCH Consulting Services:
>>> >>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> >>     > http://www.freeswitchsolutions.com
>>> >>     >
>>> >>     > 
>>> >>     > 
>>> >>     >
>>> >>     > Official FreeSWITCH Sites
>>> >>     > http://www.freeswitch.org
>>> >>     > http://wiki.freeswitch.org
>>> >>     > http://www.cluecon.com
>>> >>     >
>>> >>     > FreeSWITCH-users mailing list
>>> >>     > FreeSWITCH-users at lists.freeswitch.org
>>> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>     >
>>> >>     UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>     > http://www.freeswitch.org
>>> >>     >
>>> >>
>>> >>
>>> >>
>>> >>     --
>>> >>     Anthony Minessale II
>>> >>
>>> >>     FreeSWITCH http://www.freeswitch.org/
>>> >>     ClueCon http://www.cluecon.com/
>>> >>     Twitter: http://twitter.com/FreeSWITCH_wire
>>> >>
>>> >>     AIM: anthm
>>> >>     MSN:anthony_minessale at hotmail.com
>>> >>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> >>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> >>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>> >>
>>> >>     FreeSWITCH Developer Conference
>>> >>     sip:888 at conference.freeswitch.org
>>> >>     <mailto:sip%3A888 at conference.freeswitch.org>
>>> >>     googletalk:conf+888 at conference.freeswitch.org
>>> >>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> >>     pstn:+19193869900 <tel:%2B19193869900>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >>     Professional FreeSWITCH Consulting Services:
>>> >>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> >>     http://www.freeswitchsolutions.com
>>> >>
>>> >>     
>>> >>     
>>> >>
>>> >>     Official FreeSWITCH Sites
>>> >>     http://www.freeswitch.org
>>> >>     http://wiki.freeswitch.org
>>> >>     http://www.cluecon.com
>>> >>
>>> >>     FreeSWITCH-users mailing list
>>> >>     FreeSWITCH-users at lists.freeswitch.org
>>> >>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>     UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>     http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> ------------------------------------------------------------------------
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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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From lloyd.aloysius at gmail.com  Sat Apr 21 05:42:51 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Fri, 20 Apr 2012 21:42:51 -0400
Subject: [Freeswitch-users] Best practice for paging in an environment
 with multiple-registrations enabled on Sofia profile
In-Reply-To: <CAKV-jFyYHGdtpeKb8XWsUeidqAHAsm4EOR-WNjRb0SGy+DRoSg@mail.gmail.com>
References: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
	<4F91899D.9020104@communicatefreely.net>
	<CAKV-jFyYHGdtpeKb8XWsUeidqAHAsm4EOR-WNjRb0SGy+DRoSg@mail.gmail.com>
Message-ID: <CAGvuWYa2+jLke_n1zRn75gtYqWs2VAdGWyhcKhrDWFeyObTRkQ@mail.gmail.com>

Did you try the sofia_contact in your originate?

originate {sip_auto_answer=true} *sofia_contact*(user/2546 at domain.tld)
 &playback(alertmessage.wav)


On Fri, Apr 20, 2012 at 6:35 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Multicast isn't really the best solution, because there may be others that
> are not on the LAN.
> On Apr 20, 2012 12:07 PM, "Tim St. Pierre" <fs-list at communicatefreely.net>
> wrote:
>
>> Not to change the subject, but is multi-cast an option for you?  That
>> solves a lot of problems, and Freeswitch is capable of generating the
>> multi-cast stream.
>>
>> -Tim
>>
>> Brian Foster wrote:
>> >
>> > As the subject suggests, what is the best way to do paging with
>> > multiple-registrations enabled? In my lab I have two phones, one is
>> > registered to 2546 and 2686, the other is registered to 2546 and 2738
>> > (so both phones are registered to 2546 and another extension). I have
>> > something set up so that FS sends out a page like so:
>> >
>> > originate {sip_auto_answer=true}user/2546 at domain.tld
>> > &playback(alertmessage.wav)
>> >
>> > However as you might expect (it does make sense) the first phone to
>> > win the lottery and pick up the phone first wins the call and auto
>> > answers the call. How do I properly get every extension to playback
>> > the message at the same time?
>> >
>> > -BDF
>> >
>> > ------------------------------------------------------------------------
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Sat Apr 21 06:11:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 20 Apr 2012 22:11:10 -0400
Subject: [Freeswitch-users] Best practice for paging in an environment
 with multiple-registrations enabled on Sofia profile
In-Reply-To: <CAGvuWYa2+jLke_n1zRn75gtYqWs2VAdGWyhcKhrDWFeyObTRkQ@mail.gmail.com>
References: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
	<4F91899D.9020104@communicatefreely.net>
	<CAKV-jFyYHGdtpeKb8XWsUeidqAHAsm4EOR-WNjRb0SGy+DRoSg@mail.gmail.com>
	<CAGvuWYa2+jLke_n1zRn75gtYqWs2VAdGWyhcKhrDWFeyObTRkQ@mail.gmail.com>
Message-ID: <CAKV-jFyE9YqpgwrkfmdFbEWbwk5C=xdJOcEu81qh_V3-bQRWpw@mail.gmail.com>

I have not tried this but I did do some research on the wiki but I'm not
really sure how it world help. I'm not at my lab currently so hopefully I
can test soon. Can you elaborate why I could/would work?

-BDF
On Apr 20, 2012 9:44 PM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:

> Did you try the sofia_contact in your originate?
>
> originate {sip_auto_answer=true} *sofia_contact*(user/2546 at domain.tld)
>  &playback(alertmessage.wav)
>
>
> On Fri, Apr 20, 2012 at 6:35 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Multicast isn't really the best solution, because there may be others
>> that are not on the LAN.
>> On Apr 20, 2012 12:07 PM, "Tim St. Pierre" <fs-list at communicatefreely.net>
>> wrote:
>>
>>> Not to change the subject, but is multi-cast an option for you?  That
>>> solves a lot of problems, and Freeswitch is capable of generating the
>>> multi-cast stream.
>>>
>>> -Tim
>>>
>>> Brian Foster wrote:
>>> >
>>> > As the subject suggests, what is the best way to do paging with
>>> > multiple-registrations enabled? In my lab I have two phones, one is
>>> > registered to 2546 and 2686, the other is registered to 2546 and 2738
>>> > (so both phones are registered to 2546 and another extension). I have
>>> > something set up so that FS sends out a page like so:
>>> >
>>> > originate {sip_auto_answer=true}user/2546 at domain.tld
>>> > &playback(alertmessage.wav)
>>> >
>>> > However as you might expect (it does make sense) the first phone to
>>> > win the lottery and pick up the phone first wins the call and auto
>>> > answers the call. How do I properly get every extension to playback
>>> > the message at the same time?
>>> >
>>> > -BDF
>>> >
>>> >
>>> ------------------------------------------------------------------------
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lloyd.aloysius at gmail.com  Sat Apr 21 06:27:15 2012
From: lloyd.aloysius at gmail.com (lloyd.aloysius at gmail.com)
Date: Sat, 21 Apr 2012 02:27:15 +0000
Subject: [Freeswitch-users] Best practice for paging in an environment
	with multiple-registrations enabled on Sofia profile
In-Reply-To: <CAKV-jFyE9YqpgwrkfmdFbEWbwk5C=xdJOcEu81qh_V3-bQRWpw@mail.gmail.com>
References: <CAKV-jFwNnJw=gHWGzxiX57b2BhCTzFtT6Q1om+L4S4f69A-L4w@mail.gmail.com>
	<4F91899D.9020104@communicatefreely.net>
	<CAKV-jFyYHGdtpeKb8XWsUeidqAHAsm4EOR-WNjRb0SGy+DRoSg@mail.gmail.com>
	<CAGvuWYa2+jLke_n1zRn75gtYqWs2VAdGWyhcKhrDWFeyObTRkQ@mail.gmail.com>
	<CAKV-jFyE9YqpgwrkfmdFbEWbwk5C=xdJOcEu81qh_V3-bQRWpw@mail.gmail.com>
Message-ID: <451993939-1334975236-cardhu_decombobulator_blackberry.rim.net-1553196485-@b18.c5.bise6.blackberry>

This will work with multiple regsitrations on different phones

But not for a single phone with multiple registrations



Sent from my BlackBerry device on the Rogers Wireless Network

-----Original Message-----
From: Brian Foster <bdfoster at endigotech.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 20 Apr 2012 22:11:10 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Best practice for paging in an environment
 with multiple-registrations enabled on Sofia profile

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From philq at qsystemsengineering.com  Sat Apr 21 07:36:59 2012
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Fri, 20 Apr 2012 23:36:59 -0400
Subject: [Freeswitch-users] Possible registration/presence bug -
 Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when extension
 registration is present
Message-ID: <00ee01cd1f70$053ffc10$0fbff430$@com>

FYI, this is still a problem as of FreeSWITCH Version 1.1.beta1 (git-735e96a
2012-04-19 19-46-11 -0700).  I can generally get it to occur if I do a full
reboot of the XCP Dom0 hypervisor it's running on.  As before, restarting FS
fixes the problem.
Everything looked ok according to 'sofia status profile internal reg'.

I don't have time to figure out how to file a Jira tonight but I'll see if I
can get one filed this weekend... we're going to Hershey Park tomorrow.

Cheers,

- Phil

----------
Anthony Minessale anthony.minessale at gmail.com
<mailto:freeswitch-users%40lists.freeswitch.org?Subject=%5BFreeswitch-users%
5D%20Possible%20registration/presence%20bug%20-%0A%20Sometimes%20get%20%22Or
iginate%20Failed.%20Cause%3A%20USER_NOT_REGISTERED%22%20when%20extension%0A%
20registration%20is%20present&In-Reply-To=4F7A0B04.2070405%40integrafin.co.u
k> 
Tue Apr 3 01:02:06 MSD 2012 
*	Previous message: [Freeswitch-users] Possible registration/presence
bug - Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when
extension registration is present
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/082516.ht
ml> 
*	Next message: [Freeswitch-users] Possible registration/presence bug
- Sometimes get "Originate Failed. Cause: USER_NOT_REGISTERED" when
extension registration is present
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/082523.ht
ml> 
*	Messages sorted by: [ date ]
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/date.html
>  [ thread ]
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/thread.ht
ml>  [ subject ]
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/subject.h
tml>  [ author ]
<http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/author.ht
ml>  

I tracked this down to a race on startup where the core has not loaded
the ACL lists but sofia tries to use them for NAT detection.
I'm sure this is not something new but it should be fixed in HEAD.  (I
WISH SOMEONE WOULD HAVE OPENED A JIRA TO REFERENCE)

_____________________________________________
From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] 
Sent: Monday, April 02, 2012 3:44 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: Possible registration/presence bug - Sometimes get "Originate
Failed. Cause: USER_NOT_REGISTERED" when extension registration is present


The phone never stops working.  The problem is either present from the time
FS is started or it isn't.  I'll have to bring it down and try to get it to
fail again after hours so I can provide more detailed info.  To me, it
appears from the log I captured that presence_id is not always being
initialized properly when FS is started but I haven't looked at the source
code to see what's really happening.  I do have the two FS books on the way
though, which should give me a deeper understanding of how things work.  The
wiki only goes so far.

Also, when I do a sofia status profile internal reg, I see a bunch of long
expired registration entries.  Not sure if that could somehow be related to
this problem.

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From anita.hall at simmortel.com  Sat Apr 21 07:56:52 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Sat, 21 Apr 2012 09:26:52 +0530
Subject: [Freeswitch-users] mod_lumenvox ?
Message-ID: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>

Whatever happened to mod_lumenvox ?

regards,
Anita
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From nbhatti at gmail.com  Sat Apr 21 13:45:54 2012
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sat, 21 Apr 2012 12:45:54 +0300
Subject: [Freeswitch-users] FreeSWITCH mod_fail2ban
In-Reply-To: <4F918C17.4030909@quentustech.com>
References: <4F918C17.4030909@quentustech.com>
Message-ID: <CACak_yyd=8rN2y=dQqPAbojj2y4pMs7w-T87jgFm5dpQm01qhQ@mail.gmail.com>

This is very nice. Plain and simple. :-)

On Fri, Apr 20, 2012 at 7:17 PM, Kyle King <kyle.king at quentustech.com>wrote:

> Hello everyone, I've written a FreeSWITCH module for the application
> Fail2ban. It currently logs all inbound registration attempts, and
> failed registrations to a log file
> "/usr/local/freeswitch/log/fail2ban.log". The advantage to using this
> module
> for fail2ban rather than the standard FS log is that you don't have to
> keep your server log level as verbose.
>
> If anyone is willing to look it over and give me feedback please do so
> at https://github.com/kyconquers/freeswitch-mod_fail2ban
>
> --
> Kyle A. King
> Quentus Technologies, INC
> Cell: 703-635-9512
> Work: 253-218-6030
> Fax:  206-462-1861
> Email: kyle.king at quentustech.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gmaruzz at celliax.org  Sat Apr 21 16:37:09 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 21 Apr 2012 14:37:09 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
Message-ID: <CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>

Preferred GSM modem to use with GSMopen is Huawei E1550 dongle, or compatible.

Compatibility list (please add to this list, probably most dongles are
compatible):

    E1550
    E1552
    E169
    E1692
    E171
    E173
    E175X
    E1752C
    E1762
    E180
    K3520
    K3715
    K3765

Using more than one dongle ***REQUIRES*** the usage of an external
powered USB hub (so it can feed power to the dongles, 500mA/.5W each
one)

-giovanni

On Fri, Apr 20, 2012 at 8:32 PM, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
> Hi Giovanni,
>
> I've hit the local stores and got ZTEs, NOVIASs, SIM!s, and all sorts off
> stuf, but no Huaweys...
>
> That compatibility list would come in handy!
>
> Best,
>
> Wesley Akio
> TuntsCorp.com
>
>
>
> On Fri, Apr 20, 2012 at 2:43 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> Huawei E1550 or compatibles.
>> I'll write a list of compatible dongles maybe tomorrow.
>> -g
>>
>>
>> On 4/20/12, Ken Rice <krice at freeswitch.org> wrote:
>> > Hey Giovanni is there a good usb dongle that's widely available on
>> > somewhere
>> > like Amazon.com that you can recommend?
>> >
>> > K
>> >
>> >
>> > On 4/20/12 12:35 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>> >
>> >> On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
>> >>> Hello Giovanni,
>> >>>
>> >>> I'm off to buy one of those and will let you know how it goes...
>> >>>
>> >>> In a side note, how reliable so far?
>> >>
>> >> Hardware is very reliable and heavy duty.
>> >> Code will become better :)
>> >>
>> >>
>> >>>
>> >>> [ ]s,
>> >>>
>> >>> Wesley Akio
>> >>> TuntsCorp.com
>> >>>
>> >>>
>> >>> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
>> >>> <gmaruzz at gmail.com>wrote:
>> >>>
>> >>>> Hello FreeSWITCHers,
>> >>>>
>> >>>> the new version of GSMopen is basically usable, and the docs at the
>> >>>> wiki almost up to date (barring the details of the config file, you
>> >>>> don't need them now :) ).
>> >>>>
>> >>>> Good news is that GSMopen now supports Huawei E1550 and compatible
>> >>>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>> >>>>
>> >>>> Thanks to dongle's pcm serial streaming via USB, no more need for
>> >>>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
>> >>>> digital.
>> >>>>
>> >>>> It's reasonable to expect to be able to support many dozen concurrent
>> >>>> calls per machine (dongles are USB 2.0 compliant).
>> >>>>
>> >>>> Works well right now on Linux and Windows, and would probably works
>> >>>> on
>> >>>> *BSD and OSX with minor tweaks.
>> >>>>
>> >>>> Code will be polished, refactured, rewritten, etc. But the thing is
>> >>>> there.
>> >>>>
>> >>>> So, please update your source code to latest git, have a look at the
>> >>>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
>> >>>> begin catching bugs and documentation inaccuracies. And asking for
>> >>>> features for your use cases.
>> >>>>
>> >>>> Thanks to you all in advance, it's a pleasure and an honor to be part
>> >>>> of this community.
>> >>>>
>> >>>> http://wiki.freeswitch.org/wiki/Gsmopen
>> >>>>
>> >>>> -giovanni
>> >>>>
>> >>>>
>> >>>> --
>> >>>> Sincerely,
>> >>>>
>> >>>> Giovanni Maruzzelli
>> >>>> Cell : +39-347-2665618
>> >>>>
>> >>>>
>> >>>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> 
>> >>>> 
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://wiki.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>
>> >
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sat Apr 21 17:06:52 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 21 Apr 2012 15:06:52 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
	<CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
Message-ID: <CALXCt0pt2t2JbNT2hr+LyfUEMkgPggs6su2pqMyA3GZ_n073dA@mail.gmail.com>

Look for "Huawei modem" on Ebay or similar sites.
ZTE dongles are probably compatibles too.

Anyway, if other brands or models have the basic USB interfaces
(serial + audio) but requires different modem commands, I will add
support to them. Just contact me via mail or IRC.

-giovanni

On Sat, Apr 21, 2012 at 2:37 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Preferred GSM modem to use with GSMopen is Huawei E1550 dongle, or compatible.
>
> Compatibility list (please add to this list, probably most dongles are
> compatible):
>
> ? ?E1550
> ? ?E1552
> ? ?E169
> ? ?E1692
> ? ?E171
> ? ?E173
> ? ?E175X
> ? ?E1752C
> ? ?E1762
> ? ?E180
> ? ?K3520
> ? ?K3715
> ? ?K3765
>
> Using more than one dongle ***REQUIRES*** the usage of an external
> powered USB hub (so it can feed power to the dongles, 500mA/.5W each
> one)
>
> -giovanni
>
> On Fri, Apr 20, 2012 at 8:32 PM, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
>> Hi Giovanni,
>>
>> I've hit the local stores and got ZTEs, NOVIASs, SIM!s, and all sorts off
>> stuf, but no Huaweys...
>>
>> That compatibility list would come in handy!
>>
>> Best,
>>
>> Wesley Akio
>> TuntsCorp.com
>>
>>
>>
>> On Fri, Apr 20, 2012 at 2:43 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>> Huawei E1550 or compatibles.
>>> I'll write a list of compatible dongles maybe tomorrow.
>>> -g
>>>
>>>
>>> On 4/20/12, Ken Rice <krice at freeswitch.org> wrote:
>>> > Hey Giovanni is there a good usb dongle that's widely available on
>>> > somewhere
>>> > like Amazon.com that you can recommend?
>>> >
>>> > K
>>> >
>>> >
>>> > On 4/20/12 12:35 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>> >
>>> >> On 4/20/12, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
>>> >>> Hello Giovanni,
>>> >>>
>>> >>> I'm off to buy one of those and will let you know how it goes...
>>> >>>
>>> >>> In a side note, how reliable so far?
>>> >>
>>> >> Hardware is very reliable and heavy duty.
>>> >> Code will become better :)
>>> >>
>>> >>
>>> >>>
>>> >>> [ ]s,
>>> >>>
>>> >>> Wesley Akio
>>> >>> TuntsCorp.com
>>> >>>
>>> >>>
>>> >>> On Fri, Apr 20, 2012 at 12:58 PM, Giovanni Maruzzelli
>>> >>> <gmaruzz at gmail.com>wrote:
>>> >>>
>>> >>>> Hello FreeSWITCHers,
>>> >>>>
>>> >>>> the new version of GSMopen is basically usable, and the docs at the
>>> >>>> wiki almost up to date (barring the details of the config file, you
>>> >>>> don't need them now :) ).
>>> >>>>
>>> >>>> Good news is that GSMopen now supports Huawei E1550 and compatible
>>> >>>> dongles, that are cheap (roughly $20 on Ebay) and reliable.
>>> >>>>
>>> >>>> Thanks to dongle's pcm serial streaming via USB, no more need for
>>> >>>> sound hacks, ALSA, or any sound subsystem (and cpu load). All is
>>> >>>> digital.
>>> >>>>
>>> >>>> It's reasonable to expect to be able to support many dozen concurrent
>>> >>>> calls per machine (dongles are USB 2.0 compliant).
>>> >>>>
>>> >>>> Works well right now on Linux and Windows, and would probably works
>>> >>>> on
>>> >>>> *BSD and OSX with minor tweaks.
>>> >>>>
>>> >>>> Code will be polished, refactured, rewritten, etc. But the thing is
>>> >>>> there.
>>> >>>>
>>> >>>> So, please update your source code to latest git, have a look at the
>>> >>>> wiki page, build mod_gsmopen, connect a dongle to your machine, and
>>> >>>> begin catching bugs and documentation inaccuracies. And asking for
>>> >>>> features for your use cases.
>>> >>>>
>>> >>>> Thanks to you all in advance, it's a pleasure and an honor to be part
>>> >>>> of this community.
>>> >>>>
>>> >>>> http://wiki.freeswitch.org/wiki/Gsmopen
>>> >>>>
>>> >>>> -giovanni
>>> >>>>
>>> >>>>
>>> >>>> --
>>> >>>> Sincerely,
>>> >>>>
>>> >>>> Giovanni Maruzzelli
>>> >>>> Cell : +39-347-2665618
>>> >>>>
>>> >>>>
>>> >>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> 
>>> >>>> 
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://wiki.freeswitch.org
>>> >>>> http://www.cluecon.com
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>
>>> >
>>> >
>>> >
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveu at coppice.org  Sat Apr 21 17:53:27 2012
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 21 Apr 2012 21:53:27 +0800
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>	<CBB70921.31553%krice@freeswitch.org>	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
	<CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
Message-ID: <4F92BBD7.9040707@coppice.org>

On 04/21/2012 08:37 PM, Giovanni Maruzzelli wrote:
> Preferred GSM modem to use with GSMopen is Huawei E1550 dongle, or compatible.
>
> Compatibility list (please add to this list, probably most dongles are
> compatible):
Er, no. Compatibility depends on the chipset the dongle uses, rather 
than the actual model of the dongle. It seems some chipsets still do not 
support the audio USB endpoint which mod_gsmopen requires. A 
compatibility list will need to be draw up, model by model.
>      E1550
>      E1552
>      E169
>      E1692
>      E171
>      E173
>      E175X
>      E1752C
>      E1762
>      E180
>      K3520
>      K3715
>      K3765
>
> Using more than one dongle ***REQUIRES*** the usage of an external
> powered USB hub (so it can feed power to the dongles, 500mA/.5W each
> one)
>
> -giovanni
Steve



From gmaruzz at celliax.org  Sat Apr 21 18:02:42 2012
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 21 Apr 2012 16:02:42 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <4F92BBD7.9040707@coppice.org>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
	<CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
	<4F92BBD7.9040707@coppice.org>
Message-ID: <CALXCt0r-gAaA3sWidbSfdkpVhiKN8s69mZXzCd0XmaZTk1c6OA@mail.gmail.com>

On Sat, Apr 21, 2012 at 3:53 PM, Steve Underwood <steveu at coppice.org> wrote:

> Er, no. Compatibility depends on the chipset the dongle uses, rather
> than the actual model of the dongle. It seems some chipsets still do not
> support the audio USB endpoint which mod_gsmopen requires. A
> compatibility list will need to be draw up, model by model.

Hi Steve,
thanks for the info!

The models listed below are those that has been announced as
compatible with E1550 by the chan_dongle project, that uses the serial
and audio feature of the dongle in the same functional way as
mod_gsmopen (albeit with completely different code and logic), so the
models listed here are "kind of guaranteed" to work :).

So, let's add to the list, model by model.


>> ? ? ?E1550
>> ? ? ?E1552
>> ? ? ?E169
>> ? ? ?E1692
>> ? ? ?E171
>> ? ? ?E173
>> ? ? ?E175X
>> ? ? ?E1752C
>> ? ? ?E1762
>> ? ? ?E180
>> ? ? ?K3520
>> ? ? ?K3715
>> ? ? ?K3765
>>
>> Using more than one dongle ***REQUIRES*** the usage of an external
>> powered USB hub (so it can feed power to the dongles, 500mA/.5W each
>> one)
>>
>> -giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From rmorin at blie-ent.com  Sat Apr 21 19:01:49 2012
From: rmorin at blie-ent.com (Rob Morin)
Date: Sat, 21 Apr 2012 11:01:49 -0400
Subject: [Freeswitch-users] Dropped calls
In-Reply-To: <CAKD7_2RUyAscTys_h=z=PbLiw3zAoP0cQY6psugjC06d4MWDLQ@mail.gmail.com>
References: <11e701cd1f37$ae02e4e0$0a08aea0$@blie-ent.com>
	<CAKD7_2RUyAscTys_h=z=PbLiw3zAoP0cQY6psugjC06d4MWDLQ@mail.gmail.com>
Message-ID: <12f401cd1fcf$ae1d4ea0$0a57ebe0$@blie-ent.com>

Gill,

 

Thank you - you pointed me in the right direction.

 

I solved this by selecting "Force Timer" to Yes. The Description says this
tells it to "Use timer even when remote party does not support".  I don't
know why the difference, but at least it's working now.

 

Thank you,
Rob

 

From: Gilad Abada [mailto:ga at steadfasttelecom.com] 
Sent: Friday, April 20, 2012 5:03 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropped calls

 

Hey Rob,

 

I think you need to set your grandstreams session timers to UAC. Its on the
page where you setup the extension.

 

Hope that works

Gill

On Fri, Apr 20, 2012 at 4:53 PM, Rob Morin <rmorin at blie-ent.com> wrote:

I recently (4/17/2012) did a "make current" on my Freeswitch PBX. (Last time
I had done it was probably in January.) No particular reason for the update
- just timely.

 

After the update, we're now seeing calls drop consistently at the ~1:47
mark. Obviously, this wasn't the case beforehand.

 

I can provide a pcap file and Freeswitch log of a call demonstrating the
issue; I'm just not sure what the best method to do so is.

 

The FS server is sending a BYE, the REASON is 'SIP; cause=408; text="Session
timeout"'.

 

On my end, I have a Grandstream 502 and I can see the keep-alives are
working. But now it is getting disconnected while it wasn't before the
update.

 

Any ideas?

Thank you,

Rob

 

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





 

-- 
Gilad Abada

SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!
                 
V: 212.589.1001
F: 212.589.1011

 
For 35 years, Steadfast Telecommunications has been providing
state-of-the-art communications technology to businesses and government
agencies - large and small. Steadfast Telecommunications tailors Unified
Communications and Voice-Over IP Solutions to single-site offices or
multi-site and worldwide enterprises.   Make your virtual office a reality.
Enjoy the freedom to travel while remaining connected to your office.

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From wesleyakio at tuntscorp.com  Sat Apr 21 19:09:43 2012
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Sat, 21 Apr 2012 12:09:43 -0300
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CALXCt0r-gAaA3sWidbSfdkpVhiKN8s69mZXzCd0XmaZTk1c6OA@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
	<CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
	<4F92BBD7.9040707@coppice.org>
	<CALXCt0r-gAaA3sWidbSfdkpVhiKN8s69mZXzCd0XmaZTk1c6OA@mail.gmail.com>
Message-ID: <CAGTsDSsDtoTKJgPsm_pANKgH8+_OdJhkrmfaWGJrSmRDh5riKQ@mail.gmail.com>

I've found somewhere that the E1550 is powered by a Qualcomm MSM6246, is it
safe to assume that most(if not any) modems powered by the same chipset
should work with mod_gsmopen?


On Sat, Apr 21, 2012 at 11:02 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> On Sat, Apr 21, 2012 at 3:53 PM, Steve Underwood <steveu at coppice.org>
> wrote:
>
> > Er, no. Compatibility depends on the chipset the dongle uses, rather
> > than the actual model of the dongle. It seems some chipsets still do not
> > support the audio USB endpoint which mod_gsmopen requires. A
> > compatibility list will need to be draw up, model by model.
>
> Hi Steve,
> thanks for the info!
>
> The models listed below are those that has been announced as
> compatible with E1550 by the chan_dongle project, that uses the serial
> and audio feature of the dongle in the same functional way as
> mod_gsmopen (albeit with completely different code and logic), so the
> models listed here are "kind of guaranteed" to work :).
>
> So, let's add to the list, model by model.
>
>
> >>      E1550
> >>      E1552
> >>      E169
> >>      E1692
> >>      E171
> >>      E173
> >>      E175X
> >>      E1752C
> >>      E1762
> >>      E180
> >>      K3520
> >>      K3715
> >>      K3765
> >>
> >> Using more than one dongle ***REQUIRES*** the usage of an external
> >> powered USB hub (so it can feed power to the dongles, 500mA/.5W each
> >> one)
> >>
> >> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From gmaruzz at gmail.com  Sat Apr 21 19:19:08 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 21 Apr 2012 17:19:08 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] mod_gsmopen,
 now with USB dongle support (SMSs and Voice calls to/from Cellular
 Networks)
In-Reply-To: <CAGTsDSsDtoTKJgPsm_pANKgH8+_OdJhkrmfaWGJrSmRDh5riKQ@mail.gmail.com>
References: <CALXCt0pq2NhA1-kwCzHgEhLAeLKiknwE3Dg4YPj3Zzoo6g4Xcw@mail.gmail.com>
	<CBB70921.31553%krice@freeswitch.org>
	<CALXCt0ovv0uOLKYt3jH_4h5TDXvZ7A_7D-EwLxtfYD0VuiN10Q@mail.gmail.com>
	<CAGTsDSu1LXy=354+67ztDot_Y+_XheR-jEZEKTA7VGUpKBdnRA@mail.gmail.com>
	<CALXCt0qFFk5L2yC0cxe0twouYfLFYu4SR+rNfbdbw8BVEDDutg@mail.gmail.com>
	<4F92BBD7.9040707@coppice.org>
	<CALXCt0r-gAaA3sWidbSfdkpVhiKN8s69mZXzCd0XmaZTk1c6OA@mail.gmail.com>
	<CAGTsDSsDtoTKJgPsm_pANKgH8+_OdJhkrmfaWGJrSmRDh5riKQ@mail.gmail.com>
Message-ID: <CALXCt0rknFo5EPpwD-jWk0TeFMdZW4p_ZFwpO7oPJU2CnTrfxQ@mail.gmail.com>

On Sat, Apr 21, 2012 at 5:09 PM, Wesley Akio <wesleyakio at tuntscorp.com> wrote:
> I've found somewhere that the E1550 is powered by a?Qualcomm MSM6246, is it
> safe to assume that most(if not any) modems powered by the same chipset
> should work with mod_gsmopen?

Maybe yes, maybe not. Can be differences in firmware, or whatever.
Testing them all will tell us.
"The proof of the pudding is in the eating"

Also, I forgot to add, be sure the dongle has the "voice" capability
unlocked, or unlock it with dc-unlocker ( http://www.dc-unlocker.com/
).

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bdfoster at endigotech.com  Sat Apr 21 20:35:32 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 21 Apr 2012 12:35:32 -0400
Subject: [Freeswitch-users] mod_lumenvox ?
In-Reply-To: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>
References: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>
Message-ID: <CAKV-jFw0zZUZ75S1h01f92CbzNarA8uGNk7FyU6iUf5shWCA5w@mail.gmail.com>

Same place mod_ldap went? No clue.

-BDF

On Fri, Apr 20, 2012 at 11:56 PM, Anita Hall <anita.hall at simmortel.com>wrote:

> Whatever happened to mod_lumenvox ?
>
> regards,
> Anita
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
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From acrow at integrafin.co.uk  Sat Apr 21 22:10:25 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Sat, 21 Apr 2012 19:10:25 +0100
Subject: [Freeswitch-users] Originate call into conference - no early
 media
In-Reply-To: <CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
Message-ID: <4F92F811.6060405@integrafin.co.uk>




On 20/04/12 01:37, Anthony Minessale wrote:
> You only get it when you actually are being sent early media.
> Probably the ones that dont have any sound are only sending 180 so
> there is nothing you can do about it.
>

Anthony,

Thanks, I figured out how to fake ringback for endpoints that only send 180:

<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${pb_uuid}}loopback/*9181 
&conference(${conference_name})}"/>
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid},api_on_answer='uuid_kill 
${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
</condition>


where *9181 is an extension that does a 183 with ringback of ${uk-ring}.

Do I really need the inline=true on setting pb_uuid by the way?

The reason I didn't do the whole thing using loopback is that even when 
the parties called into the conference support wideband codecs (eg 
G722), the loopback always uses L16/8000, and so the audio is squeezed 
through a 4khz B/W channel even if the phones involved are using 
wideband codecs. If I originate into the conference I don't have that 
issue, I get HD audio all the way. Is that expected or is there a way 
around it?

eg:

<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}loopback/$1/default 
&conference(${conference_name})}"/>


Produces PSTN quality audio on dialed-in parties even though they 
negotiated G722.

Many thanks for your help

Alex




From brett at launch3.net  Sat Apr 21 22:33:00 2012
From: brett at launch3.net (Brett Wilson)
Date: Sat, 21 Apr 2012 14:33:00 -0400
Subject: [Freeswitch-users] =?utf-8?q?Group=5Fconfirm=5Fkey_on_one_channel?=
 =?utf-8?q?_only_=28using_square_brackets=29?=
Message-ID: <f810df03fccf08fa31d1e4b4b42693a5@launch3.net>

  

Hey guys, 

I am trying to do a simulataneous bridge to an internal
extension as well as a cell phone. Of course there is the issue of the
phone's voicemail picking up if the call goes too long or if the phone
is off, or the call is rejected. Group confirm key works well for this
situation however I cannot seem to get it to work with square braces,
only curly ones. I don't want confirm key to work on the internal
extension, just the cell phone. Here is a portion my dialplan: 

However
it does not work like this, only with curly braces.

Any idea how I can
accomplish this?

-- 

*******************************************

BRETT WILSON
IT DEPARTMENT
LAUNCH 3 VENTURES, LLC 
134 Myer Street

Hackensack, NJ 07601 
PHONE: 877.878.9134
FAX: 646.536.3866 
EMAIL:
Brett.Wilson at launch3.net [1]
AOL IM: Brett.Wilson at launch3.net
[2]
www.Launch3.net [3] 
www.Launch3telecom.com [4]

******************************************* 
 [5] [6] [7] [8] 


Links:
------
[1] mailto:Brett.Wilson at launch3.net
[2]
mailto:Brett.Wilson at launch3.net
[3] http://www.launch3.net/
[4]
http://www.launch3telecom.com/
[5]
http://launch3telecom.blogspot.com/
[6]
http://www.facebook.com/pages/Launch-3-Telecom/165610820178720
[7]
https://twitter.com/launch3telecom
[8]
https://plus.google.com/102713212642017290959
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From bdfoster at endigotech.com  Sun Apr 22 03:19:06 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 21 Apr 2012 19:19:06 -0400
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAKV-jFwG29ZHRFLiiPxOs0wAF-p=g3rkThnXe5RUPfF-9SQUyQ@mail.gmail.com>

Hola,

When I recieve a fax, I do something like this:

<extension name="fax">
  <condition field="destination_number" expression="fax">
    <action application="answer"/>
    <action application="sleep" data="1"/>
    <action application="set" data="fax_enable_t38_request=true"/>
    <action application="set" data="fax_enable_t38=true"/>
    <action application="set" data="execute_on_fax_success=system
/usr/local/freeswitch/scripts/success_postfax.sh '/tmp/FAX-${uuid}.tif'
'${fax_result_code} '${fax_result_text}' "/>
    <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
    <action application="hangup"/>
  </condition>
</extension>

I'm actually getting a few other fax-related variables, but I figured I
might shorten it up a bit so it wasn't extremely long on the email. Anyway,
the fax variables I'm trying to pull aren't showing up when I try and do my
post-fax processing when receiving the fax. Am I doing this all wrong?
Should I be doing something to keep the variables set by mod_spandsp alive
until I do my postfax routine?

-BDF

-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From lloyd.aloysius at gmail.com  Sun Apr 22 03:48:49 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 21 Apr 2012 19:48:49 -0400
Subject: [Freeswitch-users] Group_confirm_key on one channel only (using
 square brackets)
In-Reply-To: <f810df03fccf08fa31d1e4b4b42693a5@launch3.net>
References: <f810df03fccf08fa31d1e4b4b42693a5@launch3.net>
Message-ID: <CAGvuWYayuabHtkBagiLEethY937Mr+8O2zkUKRbt38B-qQx2Sg@mail.gmail.com>

I use the following way ... This will only prompt for the Cell Phone only

<action application="bridge"
data="{sip_invite_domain=${domain_name},ignore_early_media=true}[leg_timeout=20]user/${dialed_extension}@
${domain_name}:_:{ignore_early_media=true,group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/custom/followme/press-1-to-accept.wav,group_confirm_key=1}[leg_timeout=20]sofia/gateway/flowroute/${user_data(${dialed_extension}@${domain_name}
var followme_number)}"/>

*
*
*  <http://www.sunteltech.ca/blog/>*



On Sat, Apr 21, 2012 at 2:33 PM, Brett Wilson <brett at launch3.net> wrote:

> **
>
> Hey guys,
>
> I am trying to do a simulataneous bridge to an internal extension as well
> as a cell phone. Of course there is the issue of the phone's voicemail
> picking up if the call goes too long or if the phone is off, or the call is
> rejected. Group confirm key works well for this situation however I cannot
> seem to get it to work with square braces, only curly ones. I don't want
> confirm key to work on the internal extension, just the cell phone. Here is
> a portion my dialplan:
>
> <action application="set" data="fail_on_single_reject=true"/>
> <action application="set" data="group_confirm_cancel_timeout=1"/>
> <action application="set" data="call_timeout=25"/>
>
> <action application="bridge" data="{group_confirm_key=1,group_confirm_file=file_string://misc/eaw-customer_calling.wav!misc/eaw-accept_reject.wav}user/${user_data(101@${domain_name} attr id)},
> 		[group_confirm_key=1,group_confirm_file=file_string://misc/eaw-customer_calling.wav!misc/eaw-accept_reject.wav,ignore_early_media=true,leg_delay_start=8,leg_timeout=17,origination_caller_id_number=19735776145]sofia/gateway/flowroute/19737143925"/> <!-- Extension: 101/Front Desk -->
>
>
>
> However it does not work like this, only with curly braces.
>
>
>
> Any idea how I can accomplish this?
> 		
>
>
> --
>
> *******************************************
> *Brett Wilson*
> *IT Department*
> *Launch 3 Ventures, LLC*
> 134 Myer Street
> Hackensack, NJ 07601
> *Phone:* 877.878.9134
> *Fax:* 646.536.3866
> *Email:* Brett.Wilson at launch3.net
> *AOL IM:* Brett.Wilson at launch3.net
> www.Launch3.net <http://www.launch3.net/>
> www.Launch3telecom.com <http://www.launch3telecom.com/>
> *******************************************
> [image: Description: Description: Description: Blogger-logo]<http://launch3telecom.blogspot.com/>[image:
> Description: Description: Description: FaceBook-Logo]<http://www.facebook.com/pages/Launch-3-Telecom/165610820178720>[image:
> Description: Description: Description: Twitter-Logo]<https://twitter.com/launch3telecom>[image:
> Description: Description: Description: GPlus-Logo]<https://plus.google.com/102713212642017290959>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Sun Apr 22 05:48:04 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 21 Apr 2012 21:48:04 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKV-jFwG29ZHRFLiiPxOs0wAF-p=g3rkThnXe5RUPfF-9SQUyQ@mail.gmail.com>
References: <CAKV-jFwG29ZHRFLiiPxOs0wAF-p=g3rkThnXe5RUPfF-9SQUyQ@mail.gmail.com>
Message-ID: <CAKV-jFxuZf1RGDpqtoee4XxP-Dh6bp02esgwSeZO86pKcpBomA@mail.gmail.com>

***Message Thread Closed (no subject)***
On Apr 21, 2012 7:21 PM, "Brian Foster" <bdfoster at endigotech.com> wrote:

> Hola,
>
> When I recieve a fax, I do something like this:
>
> <extension name="fax">
>   <condition field="destination_number" expression="fax">
>     <action application="answer"/>
>     <action application="sleep" data="1"/>
>     <action application="set" data="fax_enable_t38_request=true"/>
>     <action application="set" data="fax_enable_t38=true"/>
>     <action application="set" data="execute_on_fax_success=system
> /usr/local/freeswitch/scripts/success_postfax.sh '/tmp/FAX-${uuid}.tif'
> '${fax_result_code} '${fax_result_text}' "/>
>     <action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> I'm actually getting a few other fax-related variables, but I figured I
> might shorten it up a bit so it wasn't extremely long on the email. Anyway,
> the fax variables I'm trying to pull aren't showing up when I try and do my
> post-fax processing when receiving the fax. Am I doing this all wrong?
> Should I be doing something to keep the variables set by mod_spandsp alive
> until I do my postfax routine?
>
> -BDF
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lloyd.aloysius at gmail.com  Sun Apr 22 05:48:34 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sat, 21 Apr 2012 21:48:34 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
Message-ID: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>

Hi ,

Today I update the most recent version from git. Does sofia_count_reg
 broken? It always return -1

Thanks
Lloyd
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From gabe at gundy.org  Sun Apr 22 07:46:08 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 21 Apr 2012 21:46:08 -0600
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
Message-ID: <CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>

On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
<lloyd.aloysius at gmail.com> wrote:
> Today I update the most recent?version?from git. Does?sofia_count_reg
> ?broken? It always return -1

I'll save you some time... if you think it's a bug, go here:
http://jira.freeswitch.org/secure/Dashboard.jspa

Directing users to report bugs in Jira has been a pretty constant
theme on the mailing list (especially the last few weeks).

Good luck!

Best,
Gabe


From bdfoster at endigotech.com  Sun Apr 22 08:18:12 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 22 Apr 2012 00:18:12 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
Message-ID: <CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>

Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
need to file a JIRA. This is the wrong place for bug reports.

-BDF

On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
> <lloyd.aloysius at gmail.com> wrote:
> > Today I update the most recent version from git. Does sofia_count_reg
> >  broken? It always return -1
>
> I'll save you some time... if you think it's a bug, go here:
> http://jira.freeswitch.org/secure/Dashboard.jspa
>
> Directing users to report bugs in Jira has been a pretty constant
> theme on the mailing list (especially the last few weeks).
>
> Good luck!
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From bdfoster at endigotech.com  Sun Apr 22 08:20:17 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sun, 22 Apr 2012 00:20:17 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
Message-ID: <CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>

By the way, you need to do this:

sofia_count_reg user/<user_id>@<domain>

In my case I get 0, but there's one registered. On another that's not
registered I get -1. Add that into your bug report.

-BDF

On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
> need to file a JIRA. This is the wrong place for bug reports.
>
> -BDF
>
>
> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>
>> On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
>> <lloyd.aloysius at gmail.com> wrote:
>> > Today I update the most recent version from git. Does sofia_count_reg
>> >  broken? It always return -1
>>
>> I'll save you some time... if you think it's a bug, go here:
>> http://jira.freeswitch.org/secure/Dashboard.jspa
>>
>> Directing users to report bugs in Jira has been a pretty constant
>> theme on the mailing list (especially the last few weeks).
>>
>> Good luck!
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From kyle.king at quentustech.com  Sun Apr 22 11:02:25 2012
From: kyle.king at quentustech.com (Kyle King)
Date: Sun, 22 Apr 2012 00:02:25 -0700
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
Message-ID: <4F93AD01.7010707@quentustech.com>

I tested and found everything in order. If you were using the api
without arguments then it will report a error "-1". The correct format
for counting all registrations of a user use "sofia_count_reg
<user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74" "1000"
being my user and "192.168.100.74" the IP of my FreeSWITCH box". To see
all registrations of a domain use "sofia_count_reg @<domain>" I use
"sofia_count_reg @192.168.100.74" (be sure to include the @).

Kyle A. King
Quentus Technologies, INC
Cell: 703-635-9512
Work: 253-218-6030
Fax:  206-462-1861
Email: kyle.king at quentustech.com


On 04/21/2012 09:20 PM, Brian Foster wrote:
> By the way, you need to do this:
>
> sofia_count_reg user/<user_id>@<domain>
>
> In my case I get 0, but there's one registered. On another that's not
> registered I get -1. Add that into your bug report.
>
> -BDF
>
> On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>> need to file a JIRA. This is the wrong place for bug reports.
>>
>> -BDF
>>
>>
>> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
>>> <lloyd.aloysius at gmail.com> wrote:
>>>> Today I update the most recent version from git. Does sofia_count_reg
>>>>  broken? It always return -1
>>> I'll save you some time... if you think it's a bug, go here:
>>> http://jira.freeswitch.org/secure/Dashboard.jspa
>>>
>>> Directing users to report bugs in Jira has been a pretty constant
>>> theme on the mailing list (especially the last few weeks).
>>>
>>> Good luck!
>>>
>>> Best,
>>> Gabe
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From sameer2k3t at gmail.com  Sun Apr 22 17:15:30 2012
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Sun, 22 Apr 2012 17:15:30 +0400
Subject: [Freeswitch-users] Receive chat
Message-ID: <CAA2gtHBghZ5so8Shoaxy5cu161F5y9OuBT_PZmHhUW2RVoYk-w@mail.gmail.com>

Hi,

Is there any command which can receive chat from sip channel so I can put
that received message in to a variable.
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From mario at ims.bg  Sun Apr 22 17:39:31 2012
From: mario at ims.bg (Mario Karakanovski)
Date: Sun, 22 Apr 2012 16:39:31 +0300
Subject: [Freeswitch-users] client change IP address while ringing
Message-ID: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>

Hi guys,

 

            I have an interesting situation.

I have two mobile clients and connect them using bridge. It is working
without problems, except when the called party changes its IP address
between ringing and answer.

Does anybody deal with such situation?

 

Regards,

            Mario  

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From tomp at tomp.co.uk  Sun Apr 22 20:46:16 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Sun, 22 Apr 2012 17:46:16 +0100
Subject: [Freeswitch-users] mod_httapi
Message-ID: <4F9435D8.90307@tomp.co.uk>

Hi,

Does anyone have a short guide or blog post on using the mod_httapi.

The wiki page is a bit sparse on details of how to actually use this new 
module, and have not been able to make much progress with it.

One of my main questions is what is the "action" property of the work 
element for.

The wiki says it "Change the new target url."

http://wiki.freeswitch.org/wiki/Mod_httapi

But what does that mean?

Thanks
Tom


From msc at freeswitch.org  Sun Apr 22 21:30:34 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 22 Apr 2012 10:30:34 -0700
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F9435D8.90307@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk>
Message-ID: <CAKzWOxUsEUyysEJ7JMQaJ4iVBbuihjmnAbu87v4brx2d9DyNJQ@mail.gmail.com>

This is a critical need. I have had very little time to tinker with httapi
and I never actually got that far into building an app. We definitely need
people to offer example applications with comments and a blog post/tutorial
would be a godsend. I know of someone who is working on httapi for a
commercial operation and when he's done I'm hoping to look at his work for
some inspiration so that I can add more complete examples.

If anyone has used mod_httapi and can answer Tom's question please do so.
I'm sure he'll wikify the knowledge you present. :)

-MC

On Sun, Apr 22, 2012 at 9:46 AM, Tom Parrott <tomp at tomp.co.uk> wrote:

> Hi,
>
> Does anyone have a short guide or blog post on using the mod_httapi.
>
> The wiki page is a bit sparse on details of how to actually use this new
> module, and have not been able to make much progress with it.
>
> One of my main questions is what is the "action" property of the work
> element for.
>
> The wiki says it "Change the new target url."
>
> http://wiki.freeswitch.org/wiki/Mod_httapi
>
> But what does that mean?
>
> Thanks
> Tom
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From jmesquita at freeswitch.org  Sun Apr 22 21:33:23 2012
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Sun, 22 Apr 2012 14:33:23 -0300
Subject: [Freeswitch-users] Receive chat
In-Reply-To: <CAA2gtHBghZ5so8Shoaxy5cu161F5y9OuBT_PZmHhUW2RVoYk-w@mail.gmail.com>
References: <CAA2gtHBghZ5so8Shoaxy5cu161F5y9OuBT_PZmHhUW2RVoYk-w@mail.gmail.com>
Message-ID: <CED8D28CE7DF46769868B5C53C6032D6@freeswitch.org>

I have never used it, but I assume that's what the chat plan is for, correct?

Regards,  

--  
Jo?o Mesquita
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Sunday, April 22, 2012 at 10:15 AM, Sameer Khan wrote:

> Hi,  
>  
> Is there any command which can receive chat from sip channel so I can put that received message in to a variable.
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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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From avi at avimarcus.net  Sun Apr 22 21:38:23 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 22 Apr 2012 20:38:23 +0300
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <CAKzWOxUsEUyysEJ7JMQaJ4iVBbuihjmnAbu87v4brx2d9DyNJQ@mail.gmail.com>
References: <4F9435D8.90307@tomp.co.uk>
	<CAKzWOxUsEUyysEJ7JMQaJ4iVBbuihjmnAbu87v4brx2d9DyNJQ@mail.gmail.com>
Message-ID: <CALNZuw=CK_QtsiaEOhhozt338eEw7ZHXMdN_+_=CAM-uAsWFTg@mail.gmail.com>

Not having touched it, it sounds like that means "change the URL that
future httpapi requests [for this channel?] go to." Give that a shot?

(Of note, unlike EVERY other dialplan app, lua, and other means of
controlling the dialplan, mod_httapi has security settings profiles,
limiting the permissions of what responses are possible. That means this
can be exposed to users and they can do all sorts of crazy things like pass
off who controls the script without (many) security concerns.)

-Avi


On Sun, Apr 22, 2012 at 8:30 PM, Michael Collins <msc at freeswitch.org> wrote:

> This is a critical need. I have had very little time to tinker with httapi
> and I never actually got that far into building an app. We definitely need
> people to offer example applications with comments and a blog post/tutorial
> would be a godsend. I know of someone who is working on httapi for a
> commercial operation and when he's done I'm hoping to look at his work for
> some inspiration so that I can add more complete examples.
>
> If anyone has used mod_httapi and can answer Tom's question please do so.
> I'm sure he'll wikify the knowledge you present. :)
>
> -MC
>
>
> On Sun, Apr 22, 2012 at 9:46 AM, Tom Parrott <tomp at tomp.co.uk> wrote:
>
>> Hi,
>>
>> Does anyone have a short guide or blog post on using the mod_httapi.
>>
>> The wiki page is a bit sparse on details of how to actually use this new
>> module, and have not been able to make much progress with it.
>>
>> One of my main questions is what is the "action" property of the work
>> element for.
>>
>> The wiki says it "Change the new target url."
>>
>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>
>> But what does that mean?
>>
>> Thanks
>> Tom
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From tomp at tomp.co.uk  Sun Apr 22 21:51:17 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Sun, 22 Apr 2012 18:51:17 +0100
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F9435D8.90307@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk>
Message-ID: <4F944515.609@tomp.co.uk>

Thanks Michael, yes will certainly help keep the page updated if any 
more info comes to light.

Thanks Avi, when you say "future httpapi requests" do you think that is 
when the httapi application is executed again, or is it a type of 
'callback' so that when that step in the dialplan is finished the HTTP 
API being called can instruct Freeswitch to perform another step in the 
dialplan.

Tom

On 22/04/12 17:46, Tom Parrott wrote:
> Hi,
>
> Does anyone have a short guide or blog post on using the mod_httapi.
>
> The wiki page is a bit sparse on details of how to actually use this 
> new module, and have not been able to make much progress with it.
>
> One of my main questions is what is the "action" property of the work 
> element for.
>
> The wiki says it "Change the new target url."
>
> http://wiki.freeswitch.org/wiki/Mod_httapi
>
> But what does that mean?
>
> Thanks
> Tom


From lloyd.aloysius at gmail.com  Sun Apr 22 22:14:30 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sun, 22 Apr 2012 14:14:30 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <4F93AD01.7010707@quentustech.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
Message-ID: <CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>

Environment

1. Multi-tenant
2. Core Database - mysql
3. See the results

freeswitch at default> version
FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)


freeswitch at default> sofia_count_reg aaron at mydomain.ca
-1

freeswitch at default> sofia_count_reg mydomain.ca
-1

freeswitch at default>

*
*
But the user aaron is registered with the system.




On Sun, Apr 22, 2012 at 3:02 AM, Kyle King <kyle.king at quentustech.com>wrote:

> **
> I tested and found everything in order. If you were using the api
> without arguments then it will report a error "-1". The correct format
> for counting all registrations of a user use "sofia_count_reg
> <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74" "1000"
> being my user and "192.168.100.74" the IP of my FreeSWITCH box". To see
> all registrations of a domain use "sofia_count_reg @<domain>" I use
> "sofia_count_reg @192.168.100.74" (be sure to include the @).
>
> Kyle A. King
> Quentus Technologies, INC
> Cell: 703-635-9512
> Work: 253-218-6030
> Fax:  206-462-1861
> Email: kyle.king at quentustech.com
>
>
> On 04/21/2012 09:20 PM, Brian Foster wrote:
>
> By the way, you need to do this:
>
> sofia_count_reg user/<user_id>@<domain>
>
> In my case I get 0, but there's one registered. On another that's not
> registered I get -1. Add that into your bug report.
>
> -BDF
>
> On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <bdfoster at endigotech.com>wrote:
>
>
>  Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
> need to file a JIRA. This is the wrong place for bug reports.
>
> -BDF
>
>
> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <gabe at gundy.org>wrote:
>
>
>  On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius<lloyd.aloysius at gmail.com> <lloyd.aloysius at gmail.com> wrote:
>
>  Today I update the most recent version from git. Does sofia_count_reg
>  broken? It always return -1
>
>  I'll save you some time... if you think it's a bug, go here:http://jira.freeswitch.org/secure/Dashboard.jspa
>
> Directing users to report bugs in Jira has been a pretty constant
> theme on the mailing list (especially the last few weeks).
>
> Good luck!
>
> Best,
> Gabe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>  --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From william.king at quentustech.com  Sun Apr 22 22:20:20 2012
From: william.king at quentustech.com (William King)
Date: Sun, 22 Apr 2012 11:20:20 -0700
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
	<CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
Message-ID: <4F944BE4.1050908@quentustech.com>

Can you double check how your domains are listed under 'sofia status' ?

Also, check the value for the user under the command 'sofia status
profile <profile_name> reg'. This will list all of the registrations,
and you should check the User: field. That's the domain or ip you have
to use to be able to count the registrations.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/22/2012 11:14 AM, Lloyd Aloysius wrote:
> Environment
>
> 1. Multi-tenant
> 2. Core Database - mysql
> 3. See the results
>
> freeswitch at default> version
> FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)
>
>
> freeswitch at default> sofia_count_reg aaron at mydomain.ca
> <mailto:aaron at mydomain.ca>
> -1
>
> freeswitch at default> sofia_count_reg mydomain.ca <http://mydomain.ca>
> -1
>
> freeswitch at default>
>
> *
> *
> But the user aaron is registered with the system.
>
>
>
>
> On Sun, Apr 22, 2012 at 3:02 AM, Kyle King <kyle.king at quentustech.com
> <mailto:kyle.king at quentustech.com>> wrote:
>
>     I tested and found everything in order. If you were using the api
>     without arguments then it will report a error "-1". The correct format
>     for counting all registrations of a user use "sofia_count_reg
>     <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74
>     <mailto:1000 at 192.168.100.74>" "1000"
>     being my user and "192.168.100.74" the IP of my FreeSWITCH box".
>     To see
>     all registrations of a domain use "sofia_count_reg @<domain>" I use
>     "sofia_count_reg @192.168.100.74 <http://192.168.100.74>" (be sure
>     to include the @).
>
>     Kyle A. King
>     Quentus Technologies, INC
>     Cell: 703-635-9512 <tel:703-635-9512>
>     Work: 253-218-6030 <tel:253-218-6030>
>     Fax:  206-462-1861 <tel:206-462-1861>
>     Email: kyle.king at quentustech.com <mailto:kyle.king at quentustech.com>
>
>
>     On 04/21/2012 09:20 PM, Brian Foster wrote:
>>     By the way, you need to do this:
>>
>>     sofia_count_reg user/<user_id>@<domain>
>>
>>     In my case I get 0, but there's one registered. On another that's not
>>     registered I get -1. Add that into your bug report.
>>
>>     -BDF
>>
>>     On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <mailto:bdfoster at endigotech.com>wrote:
>>
>>>     Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>>>     need to file a JIRA. This is the wrong place for bug reports.
>>>
>>>     -BDF
>>>
>>>
>>>     On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <mailto:gabe at gundy.org>wrote:
>>>
>>>>     On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
>>>>     <lloyd.aloysius at gmail.com> <mailto:lloyd.aloysius at gmail.com> wrote:
>>>>>     Today I update the most recent version from git. Does sofia_count_reg
>>>>>      broken? It always return -1
>>>>     I'll save you some time... if you think it's a bug, go here:
>>>>     http://jira.freeswitch.org/secure/Dashboard.jspa
>>>>
>>>>     Directing users to report bugs in Jira has been a pretty constant
>>>>     theme on the mailing list (especially the last few weeks).
>>>>
>>>>     Good luck!
>>>>
>>>>     Best,
>>>>     Gabe
>>>>
>>>>     _________________________________________________________________________
>>>>     Professional FreeSWITCH Consulting Services:
>>>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>     http://www.freeswitchsolutions.com
>>>>
>>>>     
>>>>     
>>>>
>>>>     Official FreeSWITCH Sites
>>>>     http://www.freeswitch.org
>>>>     http://wiki.freeswitch.org
>>>>     http://www.cluecon.com
>>>>
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>>
>>>     --
>>>     Brian D. Foster
>>>     Endigo Computer LLC
>>>     Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>>>     Phone: 317-800-7876 <tel:317-800-7876>
>>>     Indianapolis, Indiana, USA
>>>
>>>     This message contains confidential information and is intended for those
>>>     listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>>     you are not the intended recipient you are notified that disclosing,
>>>     copying, distributing or taking any action in reliance on the contents of
>>>     this information is strictly prohibited. E-mail transmission cannot be
>>>     guaranteed to be secure or error-free as information could be intercepted,
>>>     corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>>     The sender therefore does not accept liability for any errors or omissions
>>>     in the contents of this message, which arise as a result of e-mail
>>>     transmission. If verification is required please request a hard-copy
>>>     version.
>>>
>>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From kyle.king at quentustech.com  Sun Apr 22 22:32:32 2012
From: kyle.king at quentustech.com (Kyle King)
Date: Sun, 22 Apr 2012 11:32:32 -0700
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F944515.609@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
Message-ID: <4F944EC0.6070100@quentustech.com>

mod_httapi works by querying a url to obtain the xml code
to run. An example of such code is
<document type="xml/freeswitch-httapi">
<params></params>
<work>
<execute application="info"></execute>
<execute application="set" data="Kyle_test=Rock_on"></execute>
<execute application="info"></execute>
<execute application="transfer" data="9888"></execute>
</work>
</document>

Where FreeSWITCH would fist run 'info' then (if permissions are set) set
a new variable of 'Kyle_test' to the value of 'Rock_on'. FreeSWITCH
would then run 'info' again and transfer to 9888. Just replace the
execute with the action you want and confirm it is in the proper syntax
according to http://wiki.freeswitch.org/wiki/Mod_httapi
If I misunderstood stood your question and you were in fact asking how to
call this from the dial plan, then use <action application="httapi"
data="http://192.168.100.197/cgi-bin/test.cgi"/> where the url in data
is the url that will provide the xml file.

Kyle A. King
Quentus Technologies, INC
Cell: 703-635-9512
Work: 253-218-6030
Fax:  206-462-1861
Email: kyle.king at quentustech.com


On 04/22/2012 10:51 AM, Tom Parrott wrote:
> Thanks Michael, yes will certainly help keep the page updated if any 
> more info comes to light.
>
> Thanks Avi, when you say "future httpapi requests" do you think that is 
> when the httapi application is executed again, or is it a type of 
> 'callback' so that when that step in the dialplan is finished the HTTP 
> API being called can instruct Freeswitch to perform another step in the 
> dialplan.
>
> Tom
>
> On 22/04/12 17:46, Tom Parrott wrote:
>> Hi,
>>
>> Does anyone have a short guide or blog post on using the mod_httapi.
>>
>> The wiki page is a bit sparse on details of how to actually use this 
>> new module, and have not been able to make much progress with it.
>>
>> One of my main questions is what is the "action" property of the work 
>> element for.
>>
>> The wiki says it "Change the new target url."
>>
>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>
>> But what does that mean?
>>
>> Thanks
>> Tom
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From chris.chen2004 at gmail.com  Sun Apr 22 22:35:27 2012
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Sun, 22 Apr 2012 14:35:27 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
	<CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
Message-ID: <CAFEMH7MGCctnAz4Nk0n4ngMHOAi4jmf=exw+VzySq0cAQF3hqw@mail.gmail.com>

Lloyd, I have two FS, one with multi-tenant while the other single tenant,
running

version
FreeSWITCH Version 1.1.beta1 (git-a00e9af 2012-04-12 16-28-50 -0500)
FreeSWITCH Version 1.1.beta1 (git-3887495 2012-04-19 11-32-29 +0200)


both of them all returning -1 when running sofia_count_reg

I believe this should be a bug for a while.

chris



On Sun, Apr 22, 2012 at 2:14 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:

> Environment
>
> 1. Multi-tenant
> 2. Core Database - mysql
> 3. See the results
>
> freeswitch at default> version
> FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)
>
>
> freeswitch at default> sofia_count_reg aaron at mydomain.ca
> -1
>
> freeswitch at default> sofia_count_reg mydomain.ca
>  -1
>
> freeswitch at default>
>
> *
> *
> But the user aaron is registered with the system.
>
>
>
>
> On Sun, Apr 22, 2012 at 3:02 AM, Kyle King <kyle.king at quentustech.com>wrote:
>
>> **
>> I tested and found everything in order. If you were using the api
>> without arguments then it will report a error "-1". The correct format
>> for counting all registrations of a user use "sofia_count_reg
>> <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74" "1000"
>> being my user and "192.168.100.74" the IP of my FreeSWITCH box". To see
>> all registrations of a domain use "sofia_count_reg @<domain>" I use
>> "sofia_count_reg @192.168.100.74" (be sure to include the @).
>>
>> Kyle A. King
>> Quentus Technologies, INC
>> Cell: 703-635-9512
>> Work: 253-218-6030
>> Fax:  206-462-1861
>> Email: kyle.king at quentustech.com
>>
>>
>> On 04/21/2012 09:20 PM, Brian Foster wrote:
>>
>> By the way, you need to do this:
>>
>> sofia_count_reg user/<user_id>@<domain>
>>
>> In my case I get 0, but there's one registered. On another that's not
>> registered I get -1. Add that into your bug report.
>>
>> -BDF
>>
>> On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <bdfoster at endigotech.com>wrote:
>>
>>
>>  Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>> need to file a JIRA. This is the wrong place for bug reports.
>>
>> -BDF
>>
>>
>> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <gabe at gundy.org>wrote:
>>
>>
>>  On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius<lloyd.aloysius at gmail.com> <lloyd.aloysius at gmail.com> wrote:
>>
>>  Today I update the most recent version from git. Does sofia_count_reg
>>  broken? It always return -1
>>
>>  I'll save you some time... if you think it's a bug, go here:http://jira.freeswitch.org/secure/Dashboard.jspa
>>
>> Directing users to report bugs in Jira has been a pretty constant
>> theme on the mailing list (especially the last few weeks).
>>
>> Good luck!
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>  --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lloyd.aloysius at sunteltech.ca  Sun Apr 22 22:38:15 2012
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Sun, 22 Apr 2012 14:38:15 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <4F944BE4.1050908@quentustech.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
	<CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
	<4F944BE4.1050908@quentustech.com>
Message-ID: <CAGvuWYYxXb7k8KPdS6_y0qUz7trr6kb6j0U1sAEM=JnygQEx_g@mail.gmail.com>

There are no issue in domains settings. sofia_contact works perfectly
*
*
*  <http://www.sunteltech.ca/blog/>*



On Sun, Apr 22, 2012 at 2:20 PM, William King
<william.king at quentustech.com>wrote:

>  Can you double check how your domains are listed under 'sofia status' ?
>
> Also, check the value for the user under the command 'sofia status profile
> <profile_name> reg'. This will list all of the registrations, and you
> should check the User: field. That's the domain or ip you have to use to be
> able to count the registrations.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518william.king at quentustech.com
>
>
> On 04/22/2012 11:14 AM, Lloyd Aloysius wrote:
>
> Environment
>
>  1. Multi-tenant
> 2. Core Database - mysql
> 3. See the results
>
>  freeswitch at default> version
> FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)
>
>
>   freeswitch at default> sofia_count_reg aaron at mydomain.ca
> -1
>
>  freeswitch at default> sofia_count_reg mydomain.ca
>  -1
>
>  freeswitch at default>
>
>   *
> *
>  But the user aaron is registered with the system.
>
>
>
>
> On Sun, Apr 22, 2012 at 3:02 AM, Kyle King <kyle.king at quentustech.com>wrote:
>
>>  I tested and found everything in order. If you were using the api
>> without arguments then it will report a error "-1". The correct format
>> for counting all registrations of a user use "sofia_count_reg
>> <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74" "1000"
>> being my user and "192.168.100.74" the IP of my FreeSWITCH box". To see
>> all registrations of a domain use "sofia_count_reg @<domain>" I use
>> "sofia_count_reg @192.168.100.74" (be sure to include the @).
>>
>> Kyle A. King
>> Quentus Technologies, INC
>> Cell: 703-635-9512
>> Work: 253-218-6030
>> Fax:  206-462-1861
>> Email: kyle.king at quentustech.com
>>
>>
>> On 04/21/2012 09:20 PM, Brian Foster wrote:
>>
>> By the way, you need to do this:
>>
>> sofia_count_reg user/<user_id>@<domain>
>>
>> In my case I get 0, but there's one registered. On another that's not
>> registered I get -1. Add that into your bug report.
>>
>> -BDF
>>
>> On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <bdfoster at endigotech.com>wrote:
>>
>>
>>  Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>> need to file a JIRA. This is the wrong place for bug reports.
>>
>> -BDF
>>
>>
>> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <gabe at gundy.org>wrote:
>>
>>
>>  On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius<lloyd.aloysius at gmail.com> <lloyd.aloysius at gmail.com> wrote:
>>
>>  Today I update the most recent version from git. Does sofia_count_reg
>>  broken? It always return -1
>>
>>  I'll save you some time... if you think it's a bug, go here:http://jira.freeswitch.org/secure/Dashboard.jspa
>>
>> Directing users to report bugs in Jira has been a pretty constant
>> theme on the mailing list (especially the last few weeks).
>>
>> Good luck!
>>
>> Best,
>> Gabe
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>  --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From william.king at quentustech.com  Sun Apr 22 22:39:41 2012
From: william.king at quentustech.com (William King)
Date: Sun, 22 Apr 2012 11:39:41 -0700
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAFEMH7MGCctnAz4Nk0n4ngMHOAi4jmf=exw+VzySq0cAQF3hqw@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
	<CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
	<CAFEMH7MGCctnAz4Nk0n4ngMHOAi4jmf=exw+VzySq0cAQF3hqw@mail.gmail.com>
Message-ID: <4F94506D.5040001@quentustech.com>

Currently the code returns -1 if the request command syntax is invalid.

Actually this feature would be a good candidate to have auto complete
support added.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/22/2012 11:35 AM, Chris Chen wrote:
> Lloyd, I have two FS, one with multi-tenant while the other single
> tenant, running 
>
> version
> FreeSWITCH Version 1.1.beta1 (git-a00e9af 2012-04-12 16-28-50 -0500)
> FreeSWITCH Version 1.1.beta1 (git-3887495 2012-04-19 11-32-29 +0200)
>
>
> both of them all returning -1 when running sofia_count_reg
>
> I believe this should be a bug for a while.
>
> chris
>
>
>
> On Sun, Apr 22, 2012 at 2:14 PM, Lloyd Aloysius
> <lloyd.aloysius at gmail.com <mailto:lloyd.aloysius at gmail.com>> wrote:
>
>     Environment
>
>     1. Multi-tenant
>     2. Core Database - mysql
>     3. See the results
>
>     freeswitch at default> version
>     FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)
>
>
>     freeswitch at default> sofia_count_reg aaron at mydomain.ca
>     <mailto:aaron at mydomain.ca>
>     -1
>
>     freeswitch at default> sofia_count_reg mydomain.ca <http://mydomain.ca>
>     -1
>
>     freeswitch at default>
>
>     *
>     *
>     But the user aaron is registered with the system.
>
>
>
>
>     On Sun, Apr 22, 2012 at 3:02 AM, Kyle King
>     <kyle.king at quentustech.com <mailto:kyle.king at quentustech.com>> wrote:
>
>         I tested and found everything in order. If you were using the api
>         without arguments then it will report a error "-1". The
>         correct format
>         for counting all registrations of a user use "sofia_count_reg
>         <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74
>         <mailto:1000 at 192.168.100.74>" "1000"
>         being my user and "192.168.100.74" the IP of my FreeSWITCH
>         box". To see
>         all registrations of a domain use "sofia_count_reg @<domain>"
>         I use
>         "sofia_count_reg @192.168.100.74 <http://192.168.100.74>" (be
>         sure to include the @).
>
>         Kyle A. King
>         Quentus Technologies, INC
>         Cell: 703-635-9512 <tel:703-635-9512>
>         Work: 253-218-6030 <tel:253-218-6030>
>         Fax:  206-462-1861 <tel:206-462-1861>
>         Email: kyle.king at quentustech.com <mailto:kyle.king at quentustech.com>
>
>
>         On 04/21/2012 09:20 PM, Brian Foster wrote:
>>         By the way, you need to do this:
>>
>>         sofia_count_reg user/<user_id>@<domain>
>>
>>         In my case I get 0, but there's one registered. On another that's not
>>         registered I get -1. Add that into your bug report.
>>
>>         -BDF
>>
>>         On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <mailto:bdfoster at endigotech.com>wrote:
>>
>>>         Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>>>         need to file a JIRA. This is the wrong place for bug reports.
>>>
>>>         -BDF
>>>
>>>
>>>         On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <mailto:gabe at gundy.org>wrote:
>>>
>>>>         On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius
>>>>         <lloyd.aloysius at gmail.com> <mailto:lloyd.aloysius at gmail.com> wrote:
>>>>>         Today I update the most recent version from git. Does sofia_count_reg
>>>>>          broken? It always return -1
>>>>         I'll save you some time... if you think it's a bug, go here:
>>>>         http://jira.freeswitch.org/secure/Dashboard.jspa
>>>>
>>>>         Directing users to report bugs in Jira has been a pretty constant
>>>>         theme on the mailing list (especially the last few weeks).
>>>>
>>>>         Good luck!
>>>>
>>>>         Best,
>>>>         Gabe
>>>>
>>>>         _________________________________________________________________________
>>>>         Professional FreeSWITCH Consulting Services:
>>>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>         http://www.freeswitchsolutions.com
>>>>
>>>>         
>>>>         
>>>>
>>>>         Official FreeSWITCH Sites
>>>>         http://www.freeswitch.org
>>>>         http://wiki.freeswitch.org
>>>>         http://www.cluecon.com
>>>>
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org
>>>>
>>>         --
>>>         Brian D. Foster
>>>         Endigo Computer LLC
>>>         Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>>>         Phone: 317-800-7876 <tel:317-800-7876>
>>>         Indianapolis, Indiana, USA
>>>
>>>         This message contains confidential information and is intended for those
>>>         listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>>         you are not the intended recipient you are notified that disclosing,
>>>         copying, distributing or taking any action in reliance on the contents of
>>>         this information is strictly prohibited. E-mail transmission cannot be
>>>         guaranteed to be secure or error-free as information could be intercepted,
>>>         corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>>         The sender therefore does not accept liability for any errors or omissions
>>>         in the contents of this message, which arise as a result of e-mail
>>>         transmission. If verification is required please request a hard-copy
>>>         version.
>>>
>>>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From lloyd.aloysius at gmail.com  Sun Apr 22 22:40:56 2012
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sun, 22 Apr 2012 14:40:56 -0400
Subject: [Freeswitch-users] sofia_count_reg - return -1
In-Reply-To: <CAFEMH7MGCctnAz4Nk0n4ngMHOAi4jmf=exw+VzySq0cAQF3hqw@mail.gmail.com>
References: <CAGvuWYZ46DrnCx06e7A8xTTAc8L-Gch5a7v1T-9mv0yivMbEEw@mail.gmail.com>
	<CAMwZ-tJvBUons31OdqmvuVJb5b8Ng9Ub6JRiwRK-S3UVhdtBWg@mail.gmail.com>
	<CAKV-jFwUFx6YNQTaUQDSytpcX2Ct3Mm-ug1O=H6hJN7GUP=VrA@mail.gmail.com>
	<CAKV-jFyxE0oGt2gpQaDpZWU=vnLL8DK8OMjsCci1HWzBUaFPVA@mail.gmail.com>
	<4F93AD01.7010707@quentustech.com>
	<CAGvuWYbEDwhtdrRbSCs6bUxJWVRsoaow1RHWasawmA05DNoobg@mail.gmail.com>
	<CAFEMH7MGCctnAz4Nk0n4ngMHOAi4jmf=exw+VzySq0cAQF3hqw@mail.gmail.com>
Message-ID: <CAGvuWYZU-CxLOj3VT5Yvi+6WqM7+b2SvCWuvJzmfytkAM+G8ug@mail.gmail.com>

Yes. I agree. I will open this issue in JIRA.

sofia_contact works perfectly and return the value.


On Sun, Apr 22, 2012 at 2:35 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:

> Lloyd, I have two FS, one with multi-tenant while the other single tenant,
> running
>
> version
> FreeSWITCH Version 1.1.beta1 (git-a00e9af 2012-04-12 16-28-50 -0500)
> FreeSWITCH Version 1.1.beta1 (git-3887495 2012-04-19 11-32-29 +0200)
>
>
> both of them all returning -1 when running sofia_count_reg
>
> I believe this should be a bug for a while.
>
> chris
>
>
>
> On Sun, Apr 22, 2012 at 2:14 PM, Lloyd Aloysius <lloyd.aloysius at gmail.com>wrote:
>
>> Environment
>>
>> 1. Multi-tenant
>> 2. Core Database - mysql
>> 3. See the results
>>
>> freeswitch at default> version
>> FreeSWITCH Version 1.1.beta1 (git-adae5e5 2012-04-21 16-08-32 +0200)
>>
>>
>> freeswitch at default> sofia_count_reg aaron at mydomain.ca
>> -1
>>
>> freeswitch at default> sofia_count_reg mydomain.ca
>>  -1
>>
>> freeswitch at default>
>>
>> *
>> *
>> But the user aaron is registered with the system.
>>
>>
>>
>>
>> On Sun, Apr 22, 2012 at 3:02 AM, Kyle King <kyle.king at quentustech.com>wrote:
>>
>>> **
>>> I tested and found everything in order. If you were using the api
>>> without arguments then it will report a error "-1". The correct format
>>> for counting all registrations of a user use "sofia_count_reg
>>> <user>@<domain>" I use "sofia_count_reg 1000 at 192.168.100.74" "1000"
>>> being my user and "192.168.100.74" the IP of my FreeSWITCH box". To see
>>> all registrations of a domain use "sofia_count_reg @<domain>" I use
>>> "sofia_count_reg @192.168.100.74" (be sure to include the @).
>>>
>>> Kyle A. King
>>> Quentus Technologies, INC
>>> Cell: 703-635-9512
>>> Work: 253-218-6030
>>> Fax:  206-462-1861
>>> Email: kyle.king at quentustech.com
>>>
>>>
>>> On 04/21/2012 09:20 PM, Brian Foster wrote:
>>>
>>> By the way, you need to do this:
>>>
>>> sofia_count_reg user/<user_id>@<domain>
>>>
>>> In my case I get 0, but there's one registered. On another that's not
>>> registered I get -1. Add that into your bug report.
>>>
>>> -BDF
>>>
>>> On Sun, Apr 22, 2012 at 12:18 AM, Brian Foster <bdfoster at endigotech.com> <bdfoster at endigotech.com>wrote:
>>>
>>>
>>>  Looks like it's been broken for awhile now, but yes, as Gabe mentioned you
>>> need to file a JIRA. This is the wrong place for bug reports.
>>>
>>> -BDF
>>>
>>>
>>> On Sat, Apr 21, 2012 at 11:46 PM, Gabriel Gunderson <gabe at gundy.org> <gabe at gundy.org>wrote:
>>>
>>>
>>>  On Sat, Apr 21, 2012 at 7:48 PM, Lloyd Aloysius<lloyd.aloysius at gmail.com> <lloyd.aloysius at gmail.com> wrote:
>>>
>>>  Today I update the most recent version from git. Does sofia_count_reg
>>>  broken? It always return -1
>>>
>>>  I'll save you some time... if you think it's a bug, go here:http://jira.freeswitch.org/secure/Dashboard.jspa
>>>
>>> Directing users to report bugs in Jira has been a pretty constant
>>> theme on the mailing list (especially the last few weeks).
>>>
>>> Good luck!
>>>
>>> Best,
>>> Gabe
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>  --
>>> Brian D. Foster
>>> Endigo Computer LLC
>>> Email: bdfoster at endigotech.com
>>> Phone: 317-800-7876
>>> Indianapolis, Indiana, USA
>>>
>>> This message contains confidential information and is intended for those
>>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>>> you are not the intended recipient you are notified that disclosing,
>>> copying, distributing or taking any action in reliance on the contents of
>>> this information is strictly prohibited. E-mail transmission cannot be
>>> guaranteed to be secure or error-free as information could be intercepted,
>>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>>> The sender therefore does not accept liability for any errors or omissions
>>> in the contents of this message, which arise as a result of e-mail
>>> transmission. If verification is required please request a hard-copy
>>> version.
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Stefan.Weigel at allianz-warranty.com  Mon Apr 23 17:14:24 2012
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Mon, 23 Apr 2012 15:14:24 +0200
Subject: [Freeswitch-users] Mixing dynamic (xml_curl) and static
	(XML)	dialplans
In-Reply-To: <CAKbxfG-YBy0Kyn-GuVE_pXFXvZHS2bu-s5-xKu5eGS=D1cHv+g@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C67BE227791@AZWSMS03.azwarranty.int>
	<CAKV-jFy4cFEnsw73TO_c8cQDqshFZkBFz7nQsZqByZtFgme3UA@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C67BE227794@AZWSMS03.azwarranty.int>
	<CAKbxfG-YBy0Kyn-GuVE_pXFXvZHS2bu-s5-xKu5eGS=D1cHv+g@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C67BE2277B3@AZWSMS03.azwarranty.int>

Hello Anthony,

sorry for my late response. Meanwhile I decided to put all extensions into database. It's more straight forward if I need to change some detail cause I want to manage the dialplan
within a web ui.


Thanks for your reply and best reagards,

Stefan

-----------------------------------------------
-----Urspr?ngliche Nachricht-----
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Freitag, 20. April 2012 02:18
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML) dialplans

Did you miss my response?



On Wed, Apr 18, 2012 at 10:11 AM, Weigel, Stefan
<Stefan.Weigel at allianz-warranty.com> wrote:
> Hi Brian,
>
>
>
> I configured xml_curl.conf.xml with the bindings (using fs_curl as dynamic
> backend).
>
>
>
> Having this content in the static file..
>
>
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
>
>
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>
>     <context name="public">
>
>
>
>      <!--
>
>       <extension name="outside_call" continue="true">
>
>         <condition>
>
>           <action application="set" data="outside_call=true"/>
>
>         </condition>
>
>       </extension>
>
>      -->
>
>
>
>     </context>
>
>   </section>
>
> </document>
>
>
>
> ..the dialplan processing continues with xml curl. But as soon as any
> <action> command Is triggered, the dialplan processing doesn't execute xml
> curl.
>
>
>
>
>
>
>
> Best regards
>
>
>
> Stefan
>
>
>
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian
> Foster
> Gesendet: Mittwoch, 18. April 2012 16:35
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Mixing dynamic (xml_curl) and static (XML)
> dialplans
>
>
>
>
>
>>>   <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>
>
>
> Not even close. Please read and re-read
> http://wiki.freeswitch.org/wiki/Mod_xml_curl and specifically look at
> bindings.
>
> -BDF
>
> On Wed, Apr 18, 2012 at 9:20 AM, Weigel, Stefan
> <Stefan.Weigel at allianz-warranty.com> wrote:
>
> Hi,
>
>
>
> I'm trying to mix static (XML file) and dynamic (mod_xml_curl) dialplan
> content.
>
> My static dialplan looks like:
>
>
>
>
>
> /tmp/test.xml:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>
> <document type="freeswitch/xml">
>
>   <section name="dialplan" description="RE Dial Plan For FreeSwitch">
>
>     <context name="public">
>
>
>
>     <extension name="unloop">
>
>       <condition field="${unroll_loops}" expression="^true$"/>
>
>       <condition field="${sip_looped_call}" expression="^true$">
>
>         <action application="deflect" data="${destination_number}"/>
>
>       </condition>
>
>     </extension>
>
>
>
>     <extension name="outside_call" continue="true">
>
>       <condition>
>
>         <action application="set" data="outside_call=true"/>
>
>       </condition>
>
>     </extension>
>
>
>
>     <extension name="call_debug" continue="true">
>
>       <condition field="${call_debug}" expression="^true$" break="never">
>
>         <action application="info"/>
>
>       </condition>
>
>     </extension>
>
>
>
>     </context>
>
>   </section>
>
> </document>
>
>
>
> I changed parameter "dialplan" in profile internal.xml:
>
>
>
>     <param name="dialplan" value="XML:/tmp/test.xml,XML"/>
>
>
>
> But when placing a call I only see that diaplan from static XML is parsed
> and executed (message "has executed the last dialplan instruction, hanging
> up.").
>
>
>
>
>
> What's going wrong ?
>
>
>
>
>
>
>
> Thanks and cheers
>
>
>
> Stefan
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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http://www.freeswitch.org


From oh2aun at gmail.com  Mon Apr 23 17:20:04 2012
From: oh2aun at gmail.com (Mikael Nordman)
Date: Mon, 23 Apr 2012 16:20:04 +0300
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAKyTn5SomTJ_nV1VdvG7ivZ5sicXB+eUk6BUhJ1zGqqE2Y7sPA@mail.gmail.com>

Hi,
I have a asr dialogue with two consecutive play_and_detect_speech calls.

The say: part is empty and the detect: part uses nuance grxml grammar.

In the second play_and_detect invocation freeswitch looses some of the
userplane data,
thus causing the speech detection to fail.

>From the userplane data forwarded by freeswitch to the Nuance NSS server,
about one second of speech is missing in the beginning of the phrase

If I add say: with some content, then all works fine.

What could cause this loss of userplane data ?

Cheers /Mike
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From cmrienzo at gmail.com  Mon Apr 23 17:33:11 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 23 Apr 2012 09:33:11 -0400
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKyTn5SomTJ_nV1VdvG7ivZ5sicXB+eUk6BUhJ1zGqqE2Y7sPA@mail.gmail.com>
References: <CAKyTn5SomTJ_nV1VdvG7ivZ5sicXB+eUk6BUhJ1zGqqE2Y7sPA@mail.gmail.com>
Message-ID: <CAD-qRkgq=c0-LUCYHZWTu9pGzLFqx3VJnrp5F3F1REvDbvaJvw@mail.gmail.com>

I have no idea.  Use silence_stream://  if you don't want anything to be
played. Set the silence to be the same length as the speech timeout.  For
example: silence_stream://5000

If you observe incorrect behavior while playing silence, then open a jira
ticket so we can track it.

Chris



On Mon, Apr 23, 2012 at 9:20 AM, Mikael Nordman <oh2aun at gmail.com> wrote:

> Hi,
> I have a asr dialogue with two consecutive play_and_detect_speech calls.
>
> The say: part is empty and the detect: part uses nuance grxml grammar.
>
> In the second play_and_detect invocation freeswitch looses some of the
> userplane data,
> thus causing the speech detection to fail.
>
> From the userplane data forwarded by freeswitch to the Nuance NSS server,
> about one second of speech is missing in the beginning of the phrase
>
> If I add say: with some content, then all works fine.
>
> What could cause this loss of userplane data ?
>
> Cheers /Mike
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From cmrienzo at gmail.com  Mon Apr 23 17:47:16 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Mon, 23 Apr 2012 09:47:16 -0400
Subject: [Freeswitch-users] unimrcp nuance grammar failed to load
In-Reply-To: <CAGKPtBuGAT5HOGpj9xu4OkH=VpHd1Pg=EE5K0VDPKLcyjkx0dg@mail.gmail.com>
References: <CAGKPtBuGAT5HOGpj9xu4OkH=VpHd1Pg=EE5K0VDPKLcyjkx0dg@mail.gmail.com>
Message-ID: <CAD-qRkgtkGN49Wfc=kYewBcU_woN1zu7zWgY6o-MpjoxX7Spnw@mail.gmail.com>

These lines indicate the problem:

2012-04-21 01:58:01.855558 [WARNING] mrcp_client_connection.c:438 (ASR-0)
Failed to Establish TCP/MRCPv2 Connection

2012-04-21 01:58:01.855558 [WARNING] mrcp_client_connection.c:527 (ASR-0)
Null MRCPv2 Connection <17 at speechrecog>

You have mod_unimrcp configured to send an offer to 10.6.20.47 with an SDP
that says the client RTP address is at 127.0.1.1.  The MRCP server at
10.6.20.47 replies with an SDP that says the server RTP address is at
127.0.0.1.  Make sure the client and server are configured with the correct
addresses.

Also, I'm not sure if Nuance Recognizer supports JSGF grammars.

Chris


On Fri, Apr 20, 2012 at 5:05 PM, Anita Hall <anita.hall at simmortel.com>wrote:

> Hi
>
> I am getting the following error when using detect_speech from dialplan
> with unimrcp, nuance speech server and nuance recognizer. I understand that
> it is the job of the unimrcp client (ie mod_unimrcp) to convert the grammar
> from JSGF to MRCP. Why do I not see any grammar text in the SIP trace ?
> What could be wrong here ?
>
> 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:3433 (ASR-0) grammar
> failed to load, status code = 407
> 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:1541 (ASR-0) PROCESSING
> ==> ERROR
>
> Complete logs are here
> http://pastebin.freeswitch.org/18900
>
>
> The SIP trace : (10.60.20.25 is FreeSWITCH and 10.60.20.47 is Nuance
> Speech Server)
> 10.60.20.25 -> 10.60.20.47  SIP/SDP Request: INVITE sip:10.60.20.47:5060,
> with session description
> 10.60.20.47 -> 10.60.20.25  SIP/SDP Status: 200 OK, with session
> description
> 10.60.20.47 -> 10.60.20.25  SIP Request: BYE sip:10.60.20.25:5090
> 10.60.20.25 -> 10.60.20.47  SIP Status: 481 Call Does Not Exist
>
>
>
> My dialplan is
>                         <action application="answer"/>
>                         <action application="detect_speech"
> data="unimrcp:nuance5-mrcp2 yesno yesno.gram"/>
>                         <action application="detect_speech" data="grammar
> yesno"/>
>                         <action application="detect_speech"
> data="grammaron yesno"/>
>                         <action application="sleep" data="500"/>
>
> My grammar is yesno.gram kept in /usr/local/freeswitch/grammar
>
> #JSGF V1.0;
> grammar example;
> <yes> = [ yes ];
> <no> = [ no ];
> public <results> = [ <yes> | <no> ];
>
>
> # cat conf/autoload_configs/unimrcp.conf.xml
> is here
> http://pastebin.freeswitch.org/18899
>
> # cat conf/mrcp_profiles/nuance-5.0-mrcp-v2.xml
> is here
> http://pastebin.freeswitch.org/18898
>
>
>
>
>
>
>
> regards,
> Anita
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anita.hall at simmortel.com  Mon Apr 23 18:41:12 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Mon, 23 Apr 2012 20:11:12 +0530
Subject: [Freeswitch-users] unimrcp nuance grammar failed to load
In-Reply-To: <CAD-qRkgtkGN49Wfc=kYewBcU_woN1zu7zWgY6o-MpjoxX7Spnw@mail.gmail.com>
References: <CAGKPtBuGAT5HOGpj9xu4OkH=VpHd1Pg=EE5K0VDPKLcyjkx0dg@mail.gmail.com>
	<CAD-qRkgtkGN49Wfc=kYewBcU_woN1zu7zWgY6o-MpjoxX7Spnw@mail.gmail.com>
Message-ID: <CAGKPtBtnbSBZH-3ML0bv3O-ESbsYFJQUPNwWBsePFRN-i+_xnw@mail.gmail.com>

Thanks a lot Chris!

It works :)

regards,
Anita



On Mon, Apr 23, 2012 at 7:17 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> These lines indicate the problem:
>
> 2012-04-21 01:58:01.855558 [WARNING] mrcp_client_connection.c:438 (ASR-0)
> Failed to Establish TCP/MRCPv2 Connection
>
> 2012-04-21 01:58:01.855558 [WARNING] mrcp_client_connection.c:527 (ASR-0)
> Null MRCPv2 Connection <17 at speechrecog>
>
> You have mod_unimrcp configured to send an offer to 10.6.20.47 with an SDP
> that says the client RTP address is at 127.0.1.1.  The MRCP server at
> 10.6.20.47 replies with an SDP that says the server RTP address is at
> 127.0.0.1.  Make sure the client and server are configured with the correct
> addresses.
>
> Also, I'm not sure if Nuance Recognizer supports JSGF grammars.
>
> Chris
>
>
> On Fri, Apr 20, 2012 at 5:05 PM, Anita Hall <anita.hall at simmortel.com>wrote:
>
>> Hi
>>
>> I am getting the following error when using detect_speech from dialplan
>> with unimrcp, nuance speech server and nuance recognizer. I understand that
>> it is the job of the unimrcp client (ie mod_unimrcp) to convert the grammar
>> from JSGF to MRCP. Why do I not see any grammar text in the SIP trace ?
>> What could be wrong here ?
>>
>> 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:3433 (ASR-0) grammar
>> failed to load, status code = 407
>> 2012-04-21 01:58:01.855558 [DEBUG] mod_unimrcp.c:1541 (ASR-0) PROCESSING
>> ==> ERROR
>>
>> Complete logs are here
>> http://pastebin.freeswitch.org/18900
>>
>>
>> The SIP trace : (10.60.20.25 is FreeSWITCH and 10.60.20.47 is Nuance
>> Speech Server)
>> 10.60.20.25 -> 10.60.20.47  SIP/SDP Request: INVITE sip:10.60.20.47:5060,
>> with session description
>> 10.60.20.47 -> 10.60.20.25  SIP/SDP Status: 200 OK, with session
>> description
>> 10.60.20.47 -> 10.60.20.25  SIP Request: BYE sip:10.60.20.25:5090
>> 10.60.20.25 -> 10.60.20.47  SIP Status: 481 Call Does Not Exist
>>
>>
>>
>> My dialplan is
>>                         <action application="answer"/>
>>                         <action application="detect_speech"
>> data="unimrcp:nuance5-mrcp2 yesno yesno.gram"/>
>>                         <action application="detect_speech" data="grammar
>> yesno"/>
>>                         <action application="detect_speech"
>> data="grammaron yesno"/>
>>                         <action application="sleep" data="500"/>
>>
>> My grammar is yesno.gram kept in /usr/local/freeswitch/grammar
>>
>> #JSGF V1.0;
>> grammar example;
>> <yes> = [ yes ];
>> <no> = [ no ];
>> public <results> = [ <yes> | <no> ];
>>
>>
>> # cat conf/autoload_configs/unimrcp.conf.xml
>> is here
>> http://pastebin.freeswitch.org/18899
>>
>> # cat conf/mrcp_profiles/nuance-5.0-mrcp-v2.xml
>> is here
>> http://pastebin.freeswitch.org/18898
>>
>>
>>
>>
>>
>>
>>
>> regards,
>> Anita
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Mon Apr 23 19:09:46 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 23 Apr 2012 10:09:46 -0500
Subject: [Freeswitch-users] Originate call into conference - no early
	media
In-Reply-To: <4F92F811.6060405@integrafin.co.uk>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F92F811.6060405@integrafin.co.uk>
Message-ID: <CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>

If you originate loopback channels in that manner it chooses raw PCM
(L16) because it has no idea what other codec to use.
You want to avoid loopback unless its absolutely necessary since it
adds extra resource consumption dynamically allocating the audio for
each frame.

I suggest you study the books we have (not a plug, just that's where
its best documented) To get the full descriptions of how the dialplan
and codec stuff works.  It's very complicated and hard to describe in
a simple reply.



On Sat, Apr 21, 2012 at 1:10 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
>
>
>
> On 20/04/12 01:37, Anthony Minessale wrote:
>> You only get it when you actually are being sent early media.
>> Probably the ones that dont have any sound are only sending 180 so
>> there is nothing you can do about it.
>>
>
> Anthony,
>
> Thanks, I figured out how to fake ringback for endpoints that only send 180:
>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${pb_uuid}}loopback/*9181
> &conference(${conference_name})}"/>
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> ${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
> </condition>
>
>
> where *9181 is an extension that does a 183 with ringback of ${uk-ring}.
>
> Do I really need the inline=true on setting pb_uuid by the way?
>
> The reason I didn't do the whole thing using loopback is that even when
> the parties called into the conference support wideband codecs (eg
> G722), the loopback always uses L16/8000, and so the audio is squeezed
> through a 4khz B/W channel even if the phones involved are using
> wideband codecs. If I originate into the conference I don't have that
> issue, I get HD audio all the way. Is that expected or is there a way
> around it?
>
> eg:
>
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}loopback/$1/default
> &conference(${conference_name})}"/>
>
>
> Produces PSTN quality audio on dialed-in parties even though they
> negotiated G722.
>
> Many thanks for your help
>
> Alex
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From tomp at tomp.co.uk  Mon Apr 23 20:21:48 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Mon, 23 Apr 2012 17:21:48 +0100
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F944515.609@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
Message-ID: <4F95819C.7090502@tomp.co.uk>

Hi Kyle,

Thanks for your response. I wasn't being clear though.

I understand that

<action application="httapi"
data="http://192.168.100.197/cgi-bin/test.cgi"/>  in the dialplan will cause Freeswitch to retrieve the XML to perform work in the dialplan.

However in the wiki http://wiki.freeswitch.org/wiki/Mod_httapi
  under the "Work Tag" section, each type of work describes two attributes:

action and temp-action, described as "change the new target url" and "change url to submit to just for the next loop".

What is the "loop" it is describing here, and is this a kind of 'callback' to the test.cgi allowing the application to keep control of the dialplan after it has sent the initial response?

Thanks again
Tom


On 22/04/12 18:51, Tom Parrott wrote:
> Thanks Michael, yes will certainly help keep the page updated if any 
> more info comes to light.
>
> Thanks Avi, when you say "future httpapi requests" do you think that 
> is when the httapi application is executed again, or is it a type of 
> 'callback' so that when that step in the dialplan is finished the HTTP 
> API being called can instruct Freeswitch to perform another step in 
> the dialplan.
>
> Tom
>
> On 22/04/12 17:46, Tom Parrott wrote:
>> Hi,
>>
>> Does anyone have a short guide or blog post on using the mod_httapi.
>>
>> The wiki page is a bit sparse on details of how to actually use this 
>> new module, and have not been able to make much progress with it.
>>
>> One of my main questions is what is the "action" property of the work 
>> element for.
>>
>> The wiki says it "Change the new target url."
>>
>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>
>> But what does that mean?
>>
>> Thanks
>> Tom


From kyle.king at quentustech.com  Mon Apr 23 21:03:03 2012
From: kyle.king at quentustech.com (Kyle King)
Date: Mon, 23 Apr 2012 10:03:03 -0700
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F95819C.7090502@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
	<4F95819C.7090502@tomp.co.uk>
Message-ID: <4F958B47.90800@quentustech.com>

It changes the url queried. The temp-action changes for that use only
ie: that call. the action will change the url queried for a permanent
(as long as freeswitch is up) basis. Does this answer your question?

Kyle A. King
Quentus Technologies, INC
Cell: 703-635-9512
Work: 253-218-6030
Fax:  206-462-1861
Email: kyle.king at quentustech.com


On 04/23/2012 09:21 AM, Tom Parrott wrote:
> Hi Kyle,
>
> Thanks for your response. I wasn't being clear though.
>
> I understand that
>
> <action application="httapi"
> data="http://192.168.100.197/cgi-bin/test.cgi"/>  in the dialplan will cause Freeswitch to retrieve the XML to perform work in the dialplan.
>
> However in the wiki http://wiki.freeswitch.org/wiki/Mod_httapi
>   under the "Work Tag" section, each type of work describes two attributes:
>
> action and temp-action, described as "change the new target url" and "change url to submit to just for the next loop".
>
> What is the "loop" it is describing here, and is this a kind of 'callback' to the test.cgi allowing the application to keep control of the dialplan after it has sent the initial response?
>
> Thanks again
> Tom
>
>
> On 22/04/12 18:51, Tom Parrott wrote:
>> Thanks Michael, yes will certainly help keep the page updated if any 
>> more info comes to light.
>>
>> Thanks Avi, when you say "future httpapi requests" do you think that 
>> is when the httapi application is executed again, or is it a type of 
>> 'callback' so that when that step in the dialplan is finished the HTTP 
>> API being called can instruct Freeswitch to perform another step in 
>> the dialplan.
>>
>> Tom
>>
>> On 22/04/12 17:46, Tom Parrott wrote:
>>> Hi,
>>>
>>> Does anyone have a short guide or blog post on using the mod_httapi.
>>>
>>> The wiki page is a bit sparse on details of how to actually use this 
>>> new module, and have not been able to make much progress with it.
>>>
>>> One of my main questions is what is the "action" property of the work 
>>> element for.
>>>
>>> The wiki says it "Change the new target url."
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>>
>>> But what does that mean?
>>>
>>> Thanks
>>> Tom
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Apr 23 21:08:52 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 Apr 2012 10:08:52 -0700
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F958B47.90800@quentustech.com>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
	<4F95819C.7090502@tomp.co.uk> <4F958B47.90800@quentustech.com>
Message-ID: <CAKzWOxX5giY9_eur4L9p3P=o7+UPyvo54VZv3dEwdVT81KDtLA@mail.gmail.com>

On Mon, Apr 23, 2012 at 10:03 AM, Kyle King <kyle.king at quentustech.com>wrote:

> It changes the url queried. The temp-action changes for that use only
> ie: that call. the action will change the url queried for a permanent
> (as long as freeswitch is up) basis. Does this answer your question?
>

"that call" - does that mean "phone call" or "next HTTP request" or
something else? :)
-MC
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From acrow at integrafin.co.uk  Mon Apr 23 21:29:51 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 23 Apr 2012 18:29:51 +0100
Subject: [Freeswitch-users] Originate call into conference - no early
 media
In-Reply-To: <CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F92F811.6060405@integrafin.co.uk>
	<CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>
Message-ID: <4F95918F.4010801@integrafin.co.uk>

On 23/04/12 16:09, Anthony Minessale wrote:
> If you originate loopback channels in that manner it chooses raw PCM
> (L16) because it has no idea what other codec to use.
> You want to avoid loopback unless its absolutely necessary since it
> adds extra resource consumption dynamically allocating the audio for
> each frame.
>
> I suggest you study the books we have (not a plug, just that's where
> its best documented) To get the full descriptions of how the dialplan
> and codec stuff works.  It's very complicated and hard to describe in
> a simple reply.


Thanks Anthony,

I do have the FreeSwitch book and just got the cookbook. However I'd not 
come across any docs stating that loopback defaults to L16/8000 so far, 
so perhaps I should reread them.

Is there any reason why loopback can't use L16/16000 or higher so HD 
codecs can be of advantage? Just curious as my new dialplan seems to 
have solved the problem. For anyone else that is interested I have 
pasted it below (it is slightly adapted from the example):

<extension name="confringback">
<condition field="destination_number" expression="^confringback$">
<action application="set" data="ringback=$${uk-ring}"/>
<action application="bridge" 
data="{ignore_early_media=true}loopback/confringback_media"/>
</condition>
</extension>

<extension name="confringback_media">
<condition field="destination_number" expression="^confringback_media$">
<action application="pre_answer"/>
<action application="sleep" data="200000"/>
<action application="hangup"/>
</condition>
</extension>

<extension name="Add new OB call to conference">
<condition field="destination_number" expression="^add_a_call$">
<!-- ask caller for a number + #, collect into ${target_num} variable -->
<action application="play_and_get_digits" data="3 13 4 5000 # 
file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav 
ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
<!-- add this call to the conference -->
<action application="execute_extension" 
data="ADD_CALL_TO_CONF__${target_num}"/>
</condition>
</extension>

<extension name="Remove last OB call added to conference">
<condition field="destination_number" expression="^remove_a_call$">
<!-- remove a call from the conference -->
<action application="play_and_get_digits" data="3 13 1 5000 # 
file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav 
ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
<action application="set" data="res=${uuid_kill 
${hash(select/domain-${domain_name}/last_user_${target_num})}}"/>
<action application="set" data="res=${uuid_kill 
${hash(select/domain-${domain_name}/last_user_ring_${target_num})}}"/>
</condition>
</extension>

<extension name="add that call">
<!-- if we have a four-digit number then attempt to dial it as a user 
... -->
<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="hash" 
data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${pb_uuid}}loopback/confringback 
&conference(${conference_name})}"/>
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}sofia/gateway/om.net/$1 
&conference(${conference_name})}"/>
</condition>
<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(6\d{2})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}sofia/gateway/192.168.44.43/$1 
&conference(${conference_name})}"/>
</condition>
<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(49\d\d|5[3-4]\d\d|74\d\d|8\d{3}|0|9\d{3,13}|0\d{10,12})$" 
break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="hash" 
data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${pb_uuid}}loopback/*9181 
&conference(${conference_name})}"/>
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid},api_on_answer='uuid_kill 
${pb_uuid}'}sofia/gateway/10.10.0.2/$1 &conference(${conference_name})}"/>
</condition>
<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(3\d{2})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid}}sofia/gateway/192.168.44.15/$1 
&conference(${conference_name})}"/>
</condition>
<condition field="destination_number" 
expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
<action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
<action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
<action application="hash" 
data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
<action application="hash" 
data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
<action application="set" data="res=${bgapi originate 
{origination_uuid=${pb_uuid}}loopback/confringback 
&conference(${conference_name})}"/>
<action application="set" data="res=${bgapi originate 
{origination_uuid=${new_uuid},api_on_answer='uuid_kill 
${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
</condition>
<!-- ... otherwise inform moderator that the operation was not exactly 
successful -->
<condition field="destination_number" expression="^ADD_CALL_TO_CONF__$">
<action application="playback" data="ivr/ivr-please_try_again.wav"/>
</condition>
</extension>


So for 3 destinations (1st local extensions at 1xxx or 11xxxx, 2nd 
350-399 and 3rd 600-649) we don't need to fake it, for the other 2 
gateways we do.

Seems to work pretty well for now. Would it be worth getting Wiki write 
access to add that to the article about joining members by calling out 
from a conference?

Cheers

Alex


From acrow at integrafin.co.uk  Mon Apr 23 21:43:14 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 23 Apr 2012 18:43:14 +0100
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F958B47.90800@quentustech.com>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
	<4F95819C.7090502@tomp.co.uk> <4F958B47.90800@quentustech.com>
Message-ID: <4F9594B2.9040409@integrafin.co.uk>

Hi,

I am also confused. Under what circumstances if the URL queried again? 
Is it initiated from another request from the "work" section or does it 
happen after the "work" section has been completed, then auto-changing 
the request url from the original request from the XML dialplan? Eg to 
execute an application it's stated:

<execute application data action>*DATA*</execute>
                     : Execute a FreeSWITCH app.
ATTRS:
application         : The app to run
action              : Change url to submit to.
temp-action         : Change url to submit to. just for the next loop.
data                : Alternate source for app data
*DATA*              : The app data

But what is meant by "the url to submit to" and "the next loop"? IE what 
submits a request (and what is submitted) and what is the loop? Does the 
loop in fact continue hitting the webserver until the CGI returns a 
<break> in its <work> Tag?

Regards

Alex

On 23/04/12 18:03, Kyle King wrote:
> It changes the url queried. The temp-action changes for that use only
> ie: that call. the action will change the url queried for a permanent
> (as long as freeswitch is up) basis. Does this answer your question?
>
> Kyle A. King
> Quentus Technologies, INC
> Cell: 703-635-9512
> Work: 253-218-6030
> Fax:  206-462-1861
> Email: kyle.king at quentustech.com
>
>
> On 04/23/2012 09:21 AM, Tom Parrott wrote:
>> Hi Kyle,
>>
>> Thanks for your response. I wasn't being clear though.
>>
>> I understand that
>>
>> <action application="httapi"
>> data="http://192.168.100.197/cgi-bin/test.cgi"/>   in the dialplan will cause Freeswitch to retrieve the XML to perform work in the dialplan.
>>
>> However in the wiki http://wiki.freeswitch.org/wiki/Mod_httapi
>>    under the "Work Tag" section, each type of work describes two attributes:
>>
>> action and temp-action, described as "change the new target url" and "change url to submit to just for the next loop".
>>
>> What is the "loop" it is describing here, and is this a kind of 'callback' to the test.cgi allowing the application to keep control of the dialplan after it has sent the initial response?
>>
>> Thanks again
>> Tom
>>
>>
>> On 22/04/12 18:51, Tom Parrott wrote:
>>> Thanks Michael, yes will certainly help keep the page updated if any
>>> more info comes to light.
>>>
>>> Thanks Avi, when you say "future httpapi requests" do you think that
>>> is when the httapi application is executed again, or is it a type of
>>> 'callback' so that when that step in the dialplan is finished the HTTP
>>> API being called can instruct Freeswitch to perform another step in
>>> the dialplan.
>>>
>>> Tom
>>>
>>> On 22/04/12 17:46, Tom Parrott wrote:
>>>> Hi,
>>>>
>>>> Does anyone have a short guide or blog post on using the mod_httapi.
>>>>
>>>> The wiki page is a bit sparse on details of how to actually use this
>>>> new module, and have not been able to make much progress with it.
>>>>
>>>> One of my main questions is what is the "action" property of the work
>>>> element for.
>>>>
>>>> The wiki says it "Change the new target url."
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>>>
>>>> But what does that mean?
>>>>
>>>> Thanks
>>>> Tom
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From marketing at cluecon.com  Mon Apr 23 21:58:40 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 23 Apr 2012 10:58:40 -0700
Subject: [Freeswitch-users] FreeSWITCH News and Notes
Message-ID: <CAKzWOxXE2LK-Ggfxfh8gmq7aRqyjZAOGETPsox0AVfs3w+kRJQ@mail.gmail.com>

Greetings everyone. We have several items for your consideration this week:


   - The FreeSWITCH Community Conference
Call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25>will take
place at 1PM Eastern, 10AM Pacific this coming Wednesday April
   25. We will be having an informal discussion about executing dialplan
   applications or APIs using the "execute_on" and "api_on" family of channel
   variables. We look forward to having you there with us.


   - The authors of the FreeSWITCH Cookbook will be taking part in an
   interview on the weekly VUC call
<http://www.voipusersconference.org/>this Friday April 27.


   - Vestec still has some ASR starter kits to give away to FreeSWITCH
   users around the world. If you have need of speech recognition in your
   telephony application then contact marketing at cluecon.com and ask for
   more information. Vestec already supports languages in 10 of the largest
   economies of the world and more languages are in development.


Please make plans to support both conference calls this week. If you've
never been to either one I highly recommend checking them out. They are
very interesting. In fact, they are a small taste of what ClueCon is like:
hanging out with fellow telecom enthusiasts and having technical
discussions. It really is a lot of fun. Speaking of ClueCon, be sure to
register <http://www.cluecon.com/register/> today. Don't forget that if you
register in April you get extra entries in the giveaway drawings we have
throughout the conference.

Have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From anthony.minessale at gmail.com  Mon Apr 23 22:56:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 23 Apr 2012 13:56:39 -0500
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F9594B2.9040409@integrafin.co.uk>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
	<4F95819C.7090502@tomp.co.uk> <4F958B47.90800@quentustech.com>
	<4F9594B2.9040409@integrafin.co.uk>
Message-ID: <CAKbxfG_rFZbbKYcNNb0ob=LyyESFGGux-_6p7qSaDHB6tQdqYA@mail.gmail.com>

Every time the cgi returns work, once its completed, if the channel is
still alive it resubmits to the url again endlessly looking for more
work till you tell it to transfer away or hangup.  If the tmp-action
is present it submits to that url one time and subsequently goes back
to the original.



On Mon, Apr 23, 2012 at 12:43 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
> Hi,
>
> I am also confused. Under what circumstances if the URL queried again?
> Is it initiated from another request from the "work" section or does it
> happen after the "work" section has been completed, then auto-changing
> the request url from the original request from the XML dialplan? Eg to
> execute an application it's stated:
>
> <execute application data action>*DATA*</execute>
> ? ? ? ? ? ? ? ? ? ? : Execute a FreeSWITCH app.
> ATTRS:
> application ? ? ? ? : The app to run
> action ? ? ? ? ? ? ?: Change url to submit to.
> temp-action ? ? ? ? : Change url to submit to. just for the next loop.
> data ? ? ? ? ? ? ? ?: Alternate source for app data
> *DATA* ? ? ? ? ? ? ?: The app data
>
> But what is meant by "the url to submit to" and "the next loop"? IE what
> submits a request (and what is submitted) and what is the loop? Does the
> loop in fact continue hitting the webserver until the CGI returns a
> <break> in its <work> Tag?
>
> Regards
>
> Alex
>
> On 23/04/12 18:03, Kyle King wrote:
>> It changes the url queried. The temp-action changes for that use only
>> ie: that call. the action will change the url queried for a permanent
>> (as long as freeswitch is up) basis. Does this answer your question?
>>
>> Kyle A. King
>> Quentus Technologies, INC
>> Cell: 703-635-9512
>> Work: 253-218-6030
>> Fax: ?206-462-1861
>> Email: kyle.king at quentustech.com
>>
>>
>> On 04/23/2012 09:21 AM, Tom Parrott wrote:
>>> Hi Kyle,
>>>
>>> Thanks for your response. I wasn't being clear though.
>>>
>>> I understand that
>>>
>>> <action application="httapi"
>>> data="http://192.168.100.197/cgi-bin/test.cgi"/> ? in the dialplan will cause Freeswitch to retrieve the XML to perform work in the dialplan.
>>>
>>> However in the wiki http://wiki.freeswitch.org/wiki/Mod_httapi
>>> ? ?under the "Work Tag" section, each type of work describes two attributes:
>>>
>>> action and temp-action, described as "change the new target url" and "change url to submit to just for the next loop".
>>>
>>> What is the "loop" it is describing here, and is this a kind of 'callback' to the test.cgi allowing the application to keep control of the dialplan after it has sent the initial response?
>>>
>>> Thanks again
>>> Tom
>>>
>>>
>>> On 22/04/12 18:51, Tom Parrott wrote:
>>>> Thanks Michael, yes will certainly help keep the page updated if any
>>>> more info comes to light.
>>>>
>>>> Thanks Avi, when you say "future httpapi requests" do you think that
>>>> is when the httapi application is executed again, or is it a type of
>>>> 'callback' so that when that step in the dialplan is finished the HTTP
>>>> API being called can instruct Freeswitch to perform another step in
>>>> the dialplan.
>>>>
>>>> Tom
>>>>
>>>> On 22/04/12 17:46, Tom Parrott wrote:
>>>>> Hi,
>>>>>
>>>>> Does anyone have a short guide or blog post on using the mod_httapi.
>>>>>
>>>>> The wiki page is a bit sparse on details of how to actually use this
>>>>> new module, and have not been able to make much progress with it.
>>>>>
>>>>> One of my main questions is what is the "action" property of the work
>>>>> element for.
>>>>>
>>>>> The wiki says it "Change the new target url."
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>>>>
>>>>> But what does that mean?
>>>>>
>>>>> Thanks
>>>>> Tom
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> --
> This message is intended only for the addressee and may contain
> confidential information. ?Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London ?EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Apr 23 23:21:36 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 23 Apr 2012 14:21:36 -0500
Subject: [Freeswitch-users] Originate call into conference - no early
	media
In-Reply-To: <4F95918F.4010801@integrafin.co.uk>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F92F811.6060405@integrafin.co.uk>
	<CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>
	<4F95918F.4010801@integrafin.co.uk>
Message-ID: <CAKbxfG_-mmv+AAL=M-MVax0UGM7fjf37sRKBGioJe6gEgMShUA@mail.gmail.com>

You should definitely help on the wiki.  It's always worth it.

I did not mean that the book would teach you about L16 on loopback but
codecs and setting up calls.
Since you said you cannot understand why there is no early media in
your conference I thought you might benefit but studying the nuances
of call origination.

As for loopback, if its important enough to you to get that ring
sound, its really the only choice you have.
I check and it's easy enough to add a patch to let you specify the
initial codec so try out git HEAD and use the new variable
loopback_initial_codec you can specify it in the originate string like

originate {loopback_initial_codec=L16 at 16000h}loopback/1234 4321

Transcoding fees may apply.




On Mon, Apr 23, 2012 at 12:29 PM, Alex Crow <acrow at integrafin.co.uk> wrote:
> On 23/04/12 16:09, Anthony Minessale wrote:
>> If you originate loopback channels in that manner it chooses raw PCM
>> (L16) because it has no idea what other codec to use.
>> You want to avoid loopback unless its absolutely necessary since it
>> adds extra resource consumption dynamically allocating the audio for
>> each frame.
>>
>> I suggest you study the books we have (not a plug, just that's where
>> its best documented) To get the full descriptions of how the dialplan
>> and codec stuff works. ?It's very complicated and hard to describe in
>> a simple reply.
>
>
> Thanks Anthony,
>
> I do have the FreeSwitch book and just got the cookbook. However I'd not
> come across any docs stating that loopback defaults to L16/8000 so far,
> so perhaps I should reread them.
>
> Is there any reason why loopback can't use L16/16000 or higher so HD
> codecs can be of advantage? Just curious as my new dialplan seems to
> have solved the problem. For anyone else that is interested I have
> pasted it below (it is slightly adapted from the example):
>
> <extension name="confringback">
> <condition field="destination_number" expression="^confringback$">
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="bridge"
> data="{ignore_early_media=true}loopback/confringback_media"/>
> </condition>
> </extension>
>
> <extension name="confringback_media">
> <condition field="destination_number" expression="^confringback_media$">
> <action application="pre_answer"/>
> <action application="sleep" data="200000"/>
> <action application="hangup"/>
> </condition>
> </extension>
>
> <extension name="Add new OB call to conference">
> <condition field="destination_number" expression="^add_a_call$">
> <!-- ask caller for a number + #, collect into ${target_num} variable -->
> <action application="play_and_get_digits" data="3 13 4 5000 #
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> <!-- add this call to the conference -->
> <action application="execute_extension"
> data="ADD_CALL_TO_CONF__${target_num}"/>
> </condition>
> </extension>
>
> <extension name="Remove last OB call added to conference">
> <condition field="destination_number" expression="^remove_a_call$">
> <!-- remove a call from the conference -->
> <action application="play_and_get_digits" data="3 13 1 5000 #
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> <action application="set" data="res=${uuid_kill
> ${hash(select/domain-${domain_name}/last_user_${target_num})}}"/>
> <action application="set" data="res=${uuid_kill
> ${hash(select/domain-${domain_name}/last_user_ring_${target_num})}}"/>
> </condition>
> </extension>
>
> <extension name="add that call">
> <!-- if we have a four-digit number then attempt to dial it as a user
> ... -->
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${pb_uuid}}loopback/confringback
> &conference(${conference_name})}"/>
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1
> &conference(${conference_name})}"/>
> </condition>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(6\d{2})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.43/$1
> &conference(${conference_name})}"/>
> </condition>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(49\d\d|5[3-4]\d\d|74\d\d|8\d{3}|0|9\d{3,13}|0\d{10,12})$"
> break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${pb_uuid}}loopback/*9181
> &conference(${conference_name})}"/>
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> ${pb_uuid}'}sofia/gateway/10.10.0.2/$1 &conference(${conference_name})}"/>
> </condition>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(3\d{2})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.15/$1
> &conference(${conference_name})}"/>
> </condition>
> <condition field="destination_number"
> expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
> <action application="set" data="new_uuid=${create_uuid foo}" inline="true"/>
> <action application="set" data="pb_uuid=${create_uuid foo}" inline="true"/>
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> <action application="hash"
> data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${pb_uuid}}loopback/confringback
> &conference(${conference_name})}"/>
> <action application="set" data="res=${bgapi originate
> {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> ${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
> </condition>
> <!-- ... otherwise inform moderator that the operation was not exactly
> successful -->
> <condition field="destination_number" expression="^ADD_CALL_TO_CONF__$">
> <action application="playback" data="ivr/ivr-please_try_again.wav"/>
> </condition>
> </extension>
>
>
> So for 3 destinations (1st local extensions at 1xxx or 11xxxx, 2nd
> 350-399 and 3rd 600-649) we don't need to fake it, for the other 2
> gateways we do.
>
> Seems to work pretty well for now. Would it be worth getting Wiki write
> access to add that to the article about joining members by calling out
> from a conference?
>
> Cheers
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Apr 23 23:29:11 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 23 Apr 2012 12:29:11 -0700
Subject: [Freeswitch-users] Originate call into conference - no early
	media
In-Reply-To: <CAKbxfG_-mmv+AAL=M-MVax0UGM7fjf37sRKBGioJe6gEgMShUA@mail.gmail.com>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F92F811.6060405@integrafin.co.uk>
	<CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>
	<4F95918F.4010801@integrafin.co.uk>
	<CAKbxfG_-mmv+AAL=M-MVax0UGM7fjf37sRKBGioJe6gEgMShUA@mail.gmail.com>
Message-ID: <CAKzWOxXzpORKAge25gmCFpiNn_4Lgj4tb5fvFjWUPfbej3OwxA@mail.gmail.com>

Excellent description and cool new chan var.

Alex, this is definitely something you should put on the wiki. If you need
any help please let me know. Don't forget that we have a template that you
can use to add a channel variable:

http://wiki.freeswitch.org/wiki/Variable_skeleton

The rest of the documentation can go in
http://wiki.freeswitch.org/wiki/Mod_loopback. For bonus points you can use
the wiki {{:foo}} syntax to include the channel variable docs in the
mod_loopback page w/o actually duplicating your work. See the main channel
variables page for examples on how to do that.

-MC

On Mon, Apr 23, 2012 at 12:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You should definitely help on the wiki.  It's always worth it.
>
> I did not mean that the book would teach you about L16 on loopback but
> codecs and setting up calls.
> Since you said you cannot understand why there is no early media in
> your conference I thought you might benefit but studying the nuances
> of call origination.
>
> As for loopback, if its important enough to you to get that ring
> sound, its really the only choice you have.
> I check and it's easy enough to add a patch to let you specify the
> initial codec so try out git HEAD and use the new variable
> loopback_initial_codec you can specify it in the originate string like
>
> originate {loopback_initial_codec=L16 at 16000h}loopback/1234 4321
>
> Transcoding fees may apply.
>
>
>
>
> On Mon, Apr 23, 2012 at 12:29 PM, Alex Crow <acrow at integrafin.co.uk>
> wrote:
> > On 23/04/12 16:09, Anthony Minessale wrote:
> >> If you originate loopback channels in that manner it chooses raw PCM
> >> (L16) because it has no idea what other codec to use.
> >> You want to avoid loopback unless its absolutely necessary since it
> >> adds extra resource consumption dynamically allocating the audio for
> >> each frame.
> >>
> >> I suggest you study the books we have (not a plug, just that's where
> >> its best documented) To get the full descriptions of how the dialplan
> >> and codec stuff works.  It's very complicated and hard to describe in
> >> a simple reply.
> >
> >
> > Thanks Anthony,
> >
> > I do have the FreeSwitch book and just got the cookbook. However I'd not
> > come across any docs stating that loopback defaults to L16/8000 so far,
> > so perhaps I should reread them.
> >
> > Is there any reason why loopback can't use L16/16000 or higher so HD
> > codecs can be of advantage? Just curious as my new dialplan seems to
> > have solved the problem. For anyone else that is interested I have
> > pasted it below (it is slightly adapted from the example):
> >
> > <extension name="confringback">
> > <condition field="destination_number" expression="^confringback$">
> > <action application="set" data="ringback=$${uk-ring}"/>
> > <action application="bridge"
> > data="{ignore_early_media=true}loopback/confringback_media"/>
> > </condition>
> > </extension>
> >
> > <extension name="confringback_media">
> > <condition field="destination_number" expression="^confringback_media$">
> > <action application="pre_answer"/>
> > <action application="sleep" data="200000"/>
> > <action application="hangup"/>
> > </condition>
> > </extension>
> >
> > <extension name="Add new OB call to conference">
> > <condition field="destination_number" expression="^add_a_call$">
> > <!-- ask caller for a number + #, collect into ${target_num} variable -->
> > <action application="play_and_get_digits" data="3 13 4 5000 #
> >
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> > <!-- add this call to the conference -->
> > <action application="execute_extension"
> > data="ADD_CALL_TO_CONF__${target_num}"/>
> > </condition>
> > </extension>
> >
> > <extension name="Remove last OB call added to conference">
> > <condition field="destination_number" expression="^remove_a_call$">
> > <!-- remove a call from the conference -->
> > <action application="play_and_get_digits" data="3 13 1 5000 #
> >
> file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
> > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
> > <action application="set" data="res=${uuid_kill
> > ${hash(select/domain-${domain_name}/last_user_${target_num})}}"/>
> > <action application="set" data="res=${uuid_kill
> > ${hash(select/domain-${domain_name}/last_user_ring_${target_num})}}"/>
> > </condition>
> > </extension>
> >
> > <extension name="add that call">
> > <!-- if we have a four-digit number then attempt to dial it as a user
> > ... -->
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/confringback
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(6\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.43/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> >
> expression="^ADD_CALL_TO_CONF__(49\d\d|5[3-4]\d\d|74\d\d|8\d{3}|0|9\d{3,13}|0\d{10,12})$"
> > break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/*9181
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> > ${pb_uuid}'}sofia/gateway/10.10.0.2/$1&conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(3\d{2})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid}}sofia/gateway/192.168.44.15/$1
> > &conference(${conference_name})}"/>
> > </condition>
> > <condition field="destination_number"
> > expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
> > <action application="set" data="new_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="set" data="pb_uuid=${create_uuid foo}"
> inline="true"/>
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
> > <action application="hash"
> > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${pb_uuid}}loopback/confringback
> > &conference(${conference_name})}"/>
> > <action application="set" data="res=${bgapi originate
> > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
> > ${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
> > </condition>
> > <!-- ... otherwise inform moderator that the operation was not exactly
> > successful -->
> > <condition field="destination_number" expression="^ADD_CALL_TO_CONF__$">
> > <action application="playback" data="ivr/ivr-please_try_again.wav"/>
> > </condition>
> > </extension>
> >
> >
> > So for 3 destinations (1st local extensions at 1xxx or 11xxxx, 2nd
> > 350-399 and 3rd 600-649) we don't need to fake it, for the other 2
> > gateways we do.
> >
> > Seems to work pretty well for now. Would it be worth getting Wiki write
> > access to add that to the article about joining members by calling out
> > from a conference?
> >
> > Cheers
> >
> > Alex
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From acrow at integrafin.co.uk  Tue Apr 24 00:05:39 2012
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 23 Apr 2012 21:05:39 +0100
Subject: [Freeswitch-users] {Disarmed} Re: Originate call into
 conference - no early media
In-Reply-To: <CAKzWOxXzpORKAge25gmCFpiNn_4Lgj4tb5fvFjWUPfbej3OwxA@mail.gmail.com>
References: <4F8F0678.6060906@integrafin.co.uk>
	<CAKbxfG9KrJ7VrQmyAg6QdaYAa0CrjK5uOg9mox=PZFnXtapogg@mail.gmail.com>
	<4F92F811.6060405@integrafin.co.uk>
	<CAKbxfG9G2G5Xif5VBvqNPzkRnaO3p74=bbETOakKYcQjWM1vfw@mail.gmail.com>
	<4F95918F.4010801@integrafin.co.uk>
	<CAKbxfG_-mmv+AAL=M-MVax0UGM7fjf37sRKBGioJe6gEgMShUA@mail.gmail.com>
	<CAKzWOxXzpORKAge25gmCFpiNn_4Lgj4tb5fvFjWUPfbej3OwxA@mail.gmail.com>
Message-ID: <4F95B613.7040407@integrafin.co.uk>

Michael, Anthony,

Cheers guys. Really great help and also stunned with the new var! I will 
probably c/o head this weekend to try it out.

I'm not that great on wikis (I've installed and set up MediaWiki for 
others but only ever added a couple of pages to it)  but I will try to 
make a start on at least updating the conference callout example page.

Cheers

Alex



On 23/04/12 20:29, Michael Collins wrote:
> Excellent description and cool new chan var.
>
> Alex, this is definitely something you should put on the wiki. If you 
> need any help please let me know. Don't forget that we have a template 
> that you can use to add a channel variable:
>
> http://wiki.freeswitch.org/wiki/Variable_skeleton
>
> The rest of the documentation can go in 
> http://wiki.freeswitch.org/wiki/Mod_loopback. For bonus points you can 
> use the wiki {{:foo}} syntax to include the channel variable docs in 
> the mod_loopback page w/o actually duplicating your work. See the main 
> channel variables page for examples on how to do that.
>
> -MC
>
> On Mon, Apr 23, 2012 at 12:21 PM, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
>
>     You should definitely help on the wiki.  It's always worth it.
>
>     I did not mean that the book would teach you about L16 on loopback but
>     codecs and setting up calls.
>     Since you said you cannot understand why there is no early media in
>     your conference I thought you might benefit but studying the nuances
>     of call origination.
>
>     As for loopback, if its important enough to you to get that ring
>     sound, its really the only choice you have.
>     I check and it's easy enough to add a patch to let you specify the
>     initial codec so try out git HEAD and use the new variable
>     loopback_initial_codec you can specify it in the originate string like
>
>     originate {loopback_initial_codec=L16 at 16000h}loopback/1234 4321
>
>     Transcoding fees may apply.
>
>
>
>
>     On Mon, Apr 23, 2012 at 12:29 PM, Alex Crow
>     <acrow at integrafin.co.uk <mailto:acrow at integrafin.co.uk>> wrote:
>     > On 23/04/12 16:09, Anthony Minessale wrote:
>     >> If you originate loopback channels in that manner it chooses
>     raw PCM
>     >> (L16) because it has no idea what other codec to use.
>     >> You want to avoid loopback unless its absolutely necessary since it
>     >> adds extra resource consumption dynamically allocating the
>     audio for
>     >> each frame.
>     >>
>     >> I suggest you study the books we have (not a plug, just that's
>     where
>     >> its best documented) To get the full descriptions of how the
>     dialplan
>     >> and codec stuff works.  It's very complicated and hard to
>     describe in
>     >> a simple reply.
>     >
>     >
>     > Thanks Anthony,
>     >
>     > I do have the FreeSwitch book and just got the cookbook. However
>     I'd not
>     > come across any docs stating that loopback defaults to L16/8000
>     so far,
>     > so perhaps I should reread them.
>     >
>     > Is there any reason why loopback can't use L16/16000 or higher so HD
>     > codecs can be of advantage? Just curious as my new dialplan seems to
>     > have solved the problem. For anyone else that is interested I have
>     > pasted it below (it is slightly adapted from the example):
>     >
>     > <extension name="confringback">
>     > <condition field="destination_number" expression="^confringback$">
>     > <action application="set" data="ringback=$${uk-ring}"/>
>     > <action application="bridge"
>     > data="{ignore_early_media=true}loopback/confringback_media"/>
>     > </condition>
>     > </extension>
>     >
>     > <extension name="confringback_media">
>     > <condition field="destination_number"
>     expression="^confringback_media$">
>     > <action application="pre_answer"/>
>     > <action application="sleep" data="200000"/>
>     > <action application="hangup"/>
>     > </condition>
>     > </extension>
>     >
>     > <extension name="Add new OB call to conference">
>     > <condition field="destination_number" expression="^add_a_call$">
>     > <!-- ask caller for a number + #, collect into ${target_num}
>     variable -->
>     > <action application="play_and_get_digits" data="3 13 4 5000 #
>     >
>     file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
>     > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
>     > <!-- add this call to the conference -->
>     > <action application="execute_extension"
>     > data="ADD_CALL_TO_CONF__${target_num}"/>
>     > </condition>
>     > </extension>
>     >
>     > <extension name="Remove last OB call added to conference">
>     > <condition field="destination_number" expression="^remove_a_call$">
>     > <!-- remove a call from the conference -->
>     > <action application="play_and_get_digits" data="3 13 1 5000 #
>     >
>     file_string://ivr/ivr-please_enter_the_phone_number.wav!ivr/ivr-followed_by_pound.wav
>     > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+"/>
>     > <action application="set" data="res=${uuid_kill
>     > ${hash(select/domain-${domain_name}/last_user_${target_num})}}"/>
>     > <action application="set" data="res=${uuid_kill
>     >
>     ${hash(select/domain-${domain_name}/last_user_ring_${target_num})}}"/>
>     > </condition>
>     > </extension>
>     >
>     > <extension name="add that call">
>     > <!-- if we have a four-digit number then attempt to dial it as a
>     user
>     > ... -->
>     > <condition field="destination_number"
>     > expression="^ADD_CALL_TO_CONF__(2\d{2})$" break="on-true">
>     > <action application="set" data="new_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="set" data="pb_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${pb_uuid}}loopback/confringback
>     > &conference(${conference_name})}"/>
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${new_uuid}}sofia/gateway/om.net/$1
>     <http://om.net/$1>
>     > &conference(${conference_name})}"/>
>     > </condition>
>     > <condition field="destination_number"
>     > expression="^ADD_CALL_TO_CONF__(6\d{2})$" break="on-true">
>     > <action application="set" data="new_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${new_uuid}}sofia/gateway/*MailScanner
>     warning: numerical links are often malicious:* 192.168.44.43/$1
>     <http://192.168.44.43/$1>
>     > &conference(${conference_name})}"/>
>     > </condition>
>     > <condition field="destination_number"
>     >
>     expression="^ADD_CALL_TO_CONF__(49\d\d|5[3-4]\d\d|74\d\d|8\d{3}|0|9\d{3,13}|0\d{10,12})$"
>     > break="on-true">
>     > <action application="set" data="new_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="set" data="pb_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${pb_uuid}}loopback/*9181
>     > &conference(${conference_name})}"/>
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
>     > ${pb_uuid}'}sofia/gateway/*MailScanner warning: numerical links
>     are often malicious:* 10.10.0.2/$1 <http://10.10.0.2/$1>
>     &conference(${conference_name})}"/>
>     > </condition>
>     > <condition field="destination_number"
>     > expression="^ADD_CALL_TO_CONF__(3\d{2})$" break="on-true">
>     > <action application="set" data="new_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${new_uuid}}sofia/gateway/*MailScanner
>     warning: numerical links are often malicious:* 192.168.44.15/$1
>     <http://192.168.44.15/$1>
>     > &conference(${conference_name})}"/>
>     > </condition>
>     > <condition field="destination_number"
>     > expression="^ADD_CALL_TO_CONF__(1\d{3}|11\d{4})$" break="on-true">
>     > <action application="set" data="new_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="set" data="pb_uuid=${create_uuid foo}"
>     inline="true"/>
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_$1/${new_uuid}" />
>     > <action application="hash"
>     > data="insert/domain-${domain_name}/last_user_ring_$1/${pb_uuid}" />
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${pb_uuid}}loopback/confringback
>     > &conference(${conference_name})}"/>
>     > <action application="set" data="res=${bgapi originate
>     > {origination_uuid=${new_uuid},api_on_answer='uuid_kill
>     > ${pb_uuid}'}user/$1 &conference(${conference_name})}"/>
>     > </condition>
>     > <!-- ... otherwise inform moderator that the operation was not
>     exactly
>     > successful -->
>     > <condition field="destination_number"
>     expression="^ADD_CALL_TO_CONF__$">
>     > <action application="playback" data="ivr/ivr-please_try_again.wav"/>
>     > </condition>
>     > </extension>
>     >
>     >
>     > So for 3 destinations (1st local extensions at 1xxx or 11xxxx, 2nd
>     > 350-399 and 3rd 600-649) we don't need to fake it, for the other 2
>     > gateways we do.
>     >
>     > Seems to work pretty well for now. Would it be worth getting
>     Wiki write
>     > access to add that to the article about joining members by
>     calling out
>     > from a conference?
>     >
>     > Cheers
>     >
>     > Alex
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>
>
>
>     --
>     Anthony Minessale II
>
>     FreeSWITCH http://www.freeswitch.org/
>     ClueCon http://www.cluecon.com/
>     Twitter: http://twitter.com/FreeSWITCH_wire
>
>     AIM: anthm
>     MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:+19193869900 <tel:%2B19193869900>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> -- 
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From me at nevian.org  Tue Apr 24 00:43:08 2012
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 24 Apr 2012 00:43:08 +0400
Subject: [Freeswitch-users] per user/call session_timer
Message-ID: <181771335213788@web20f.yandex.ru>

Hello,

Can I disable session timers on per user or per call basis?

-- 
wbr,
Serge


From kris at kriskinc.com  Tue Apr 24 01:46:59 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 23 Apr 2012 17:46:59 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CAKV-jFzYPKAXZL3pQuWwvQ-8uKWa61HM=cisbiypPWkQMjKXHQ@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
	<CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
	<CAKV-jFzYPKAXZL3pQuWwvQ-8uKWa61HM=cisbiypPWkQMjKXHQ@mail.gmail.com>
Message-ID: <CAKDfjgdEuM5Dx02UPmpqAVUOZ9_br8TgHVRwra8eBm-TtJVk6A@mail.gmail.com>

These are filetypes for mod_sndfile, not mod_native_file.

On Thu, Apr 19, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> AIFF (Apple/SGI) (extension "aiff")
> AU (Sun/NeXT) (extension "au")
> AVR (Audio Visual Research) (extension "avr")
> CAF (Apple Core Audio File) (extension "caf")
> HTK (HMM Tool Kit) (extension "htk")
> IFF (Amiga IFF/SVX8/SV16) (extension "iff")
> MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat")
> MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat")
> PAF (Ensoniq PARIS) (extension "paf")
> PVF (Portable Voice Format) (extension "pvf")
> RAW (header-less) (extension "raw")
> SD2 (Sound Designer II) (extension "sd2")
> SDS (Midi Sample Dump Standard) (extension "sds")
> SF (Berkeley/IRCAM/CARL) (extension "sf")
> VOC (Creative Labs) (extension "voc")
> W64 (SoundFoundry WAVE 64) (extension "w64")
> WAV (Microsoft) (extension "wav")
> WAV (NIST Sphere) (extension "wav")
> WAVEX (Microsoft) (extension "wav")
> XI (FastTracker 2) (extension "xi")
>
> Pick one, and live with it.
>
>
> -BDF
>
>
> On Thu, Apr 19, 2012 at 2:41 PM, Bernard Fluixa <fluixab at bellsouth.net>
> wrote:
>>
>> /usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX
>>
>> fs_encode log:
>> 2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File
>> [a.SPEEX] 8000hz
>> Frame size is 160
>>
>> I got the a.SPEEX file and same squelch noise when playing it.
>>
>> 2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File
>> [a.SPEEX] 8000hz
>> 2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>> Any specific parameters to pass to fs_encode or playback?
>>
>> Thanks
>>
>> Bernard
>>
>> On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:
>>
>> This sounds like a totally different issue.
>> Please try using FreeSWITCH's encoding functionality:
>>
>> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>>
>> -Avi
>>
>>
>> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> wrote:
>>>
>>> There's still something I'm not clear with. Here's what I did
>>>
>>> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
>>> 2) Call a user with x-Lite soft phone: originate user/1001
>>> &playback(/home/sounds/a.SPEEX)
>>>
>>> ==> file is open and played but one can hear only noise squelch. What am
>>> I doing wrong?
>>>
>>> FYI, I then converted a.SPEEX file back to wav format and could play that
>>> wav file.
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>>>
>>> > no you, cant modify it. ?This is a function of mod_native_file and its
>>> > limited to matching the iananames they registered the codec with.
>>> >
>>> >
>>> >
>>> > On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>> >> It seems all the codecs are are chosen in uppercase, as it seems to be
>>> >> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>>> >> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>>> >> You can probably modify mod_speex to add an alias to load .spx files
>>> >> also.
>>> >>
>>> >> -Avi
>>> >>
>>> >>
>>> >> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa
>>> >> <fluixab at bellsouth.net>
>>> >> wrote:
>>> >>>
>>> >>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension
>>> >>> FROM
>>> >>> Freeswitch and could play it with playback command. No problem.
>>> >>> However, it
>>> >>> is my understanding that speex format files have ".spx" extension.
>>> >>> Why
>>> >>> doesn't Freeswitch allows for ".spx" extension?
>>> >>>
>>> >>> .speex extension are not accepted.
>>> >>>
>>> >>> Thanks again
>>> >>>
>>> >>> Bernard
>>> >>>
>>> >>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>>> >>>
>>> >>> Didn't we answer this yesterday? Bernard, have you tried it yet with
>>> >>> a
>>> >>> .SPEEX (or .speex?) extension?
>>> >>>
>>> >>> -Avi
>>> >>>
>>> >>>
>>> >>>
>>> >>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa
>>> >>> <fluixab at bellsouth.net>
>>> >>> wrote:
>>> >>>>
>>> >>>> Hello,
>>> >>>>
>>> >>>> I cannot play ".spx" (speex) file in commands such as originate?
>>> >>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>>> >>>> Freeswitch returns
>>> >>>>
>>> >>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>>> >>>> format [spx] for [test.spx]!
>>> >>>>
>>> >>>> I converted file to wav format and could play it (speexdec test.spx
>>> >>>> test.wav) with playback command. I probably missed a point here but
>>> >>>> I would
>>> >>>> like to know how I can play speex files without having to convert
>>> >>>> them to
>>> >>>> wav first.
>>> >>>>
>>> >>>> Anyone?
>>> >>>>
>>> >>>> Thank you
>>> >>>>
>>> >>>> Bernard
>>> >>>>
>>> >>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> 
>>> >>>> 
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://wiki.freeswitch.org
>>> >>>> http://www.cluecon.com
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>
>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >>>
>>> >>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> 
>>> >>> 
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://wiki.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> 
>>> >>> 
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://wiki.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org
>>> > googletalk:conf+888 at conference.freeswitch.org
>>> > pstn:+19193869900
>>> >
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From fluixab at bellsouth.net  Tue Apr 24 01:50:49 2012
From: fluixab at bellsouth.net (Bernard Fluixa)
Date: Mon, 23 Apr 2012 17:50:49 -0400
Subject: [Freeswitch-users] Speex file format
In-Reply-To: <CAKDfjgdEuM5Dx02UPmpqAVUOZ9_br8TgHVRwra8eBm-TtJVk6A@mail.gmail.com>
References: <8B72D8AA-E7BA-4BF4-BB02-ECA796104BB8@bellsouth.net>
	<CALNZuwm8DmhvZeoHTzopdTrf8-MRHqcNK=bhDsV_wLPfL2OX0Q@mail.gmail.com>
	<16CDE330-AB6A-4F9E-9F41-74373D643016@bellsouth.net>
	<CALNZuw=ADeNqrkRdRSidL9cYVxqQwbppOVQ3zA0cSG9rEY1Ldw@mail.gmail.com>
	<CAKbxfG-HvFzb40Z-VsS5MndJHUpr_JeVjmkzVQ=7g6UhXft4Jg@mail.gmail.com>
	<BDE5F18B-B594-43AF-80B0-293D3D42579B@bellsouth.net>
	<CALNZuwk7aBgecs4FH3Y45KCeVOUhTWJ0ORqNRegSGd4t+nXrYw@mail.gmail.com>
	<CC32BC27-21D9-4208-A8CA-6DC4E74D170A@bellsouth.net>
	<CAKV-jFzYPKAXZL3pQuWwvQ-8uKWa61HM=cisbiypPWkQMjKXHQ@mail.gmail.com>
	<CAKDfjgdEuM5Dx02UPmpqAVUOZ9_br8TgHVRwra8eBm-TtJVk6A@mail.gmail.com>
Message-ID: <3C7C2019-7E89-4873-9128-DDF7B779C15E@bellsouth.net>

Already received this. Anthony already explained me.
On Apr 23, 2012, at 5:46 PM, Kristian Kielhofner wrote:

> These are filetypes for mod_sndfile, not mod_native_file.
> 
> On Thu, Apr 19, 2012 at 2:59 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>> AIFF (Apple/SGI) (extension "aiff")
>> AU (Sun/NeXT) (extension "au")
>> AVR (Audio Visual Research) (extension "avr")
>> CAF (Apple Core Audio File) (extension "caf")
>> HTK (HMM Tool Kit) (extension "htk")
>> IFF (Amiga IFF/SVX8/SV16) (extension "iff")
>> MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat")
>> MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat")
>> PAF (Ensoniq PARIS) (extension "paf")
>> PVF (Portable Voice Format) (extension "pvf")
>> RAW (header-less) (extension "raw")
>> SD2 (Sound Designer II) (extension "sd2")
>> SDS (Midi Sample Dump Standard) (extension "sds")
>> SF (Berkeley/IRCAM/CARL) (extension "sf")
>> VOC (Creative Labs) (extension "voc")
>> W64 (SoundFoundry WAVE 64) (extension "w64")
>> WAV (Microsoft) (extension "wav")
>> WAV (NIST Sphere) (extension "wav")
>> WAVEX (Microsoft) (extension "wav")
>> XI (FastTracker 2) (extension "xi")
>> 
>> Pick one, and live with it.
>> 
>> 
>> -BDF
>> 
>> 
>> On Thu, Apr 19, 2012 at 2:41 PM, Bernard Fluixa <fluixab at bellsouth.net>
>> wrote:
>>> 
>>> /usr/local/freeswitch/bin/fs_encode -l mod_speex a.wav a.SPEEX
>>> 
>>> fs_encode log:
>>> 2012-04-19 14:37:08.055049 [INFO] mod_native_file.c:94 Opening File
>>> [a.SPEEX] 8000hz
>>> Frame size is 160
>>> 
>>> I got the a.SPEEX file and same squelch noise when playing it.
>>> 
>>> 2012-04-19 14:37:19.751293 [INFO] mod_native_file.c:94 Opening File
>>> [a.SPEEX] 8000hz
>>> 2012-04-19 14:37:19.751293 [DEBUG] switch_ivr_play_say.c:1309 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>> 
>>> Any specific parameters to pass to fs_encode or playback?
>>> 
>>> Thanks
>>> 
>>> Bernard
>>> 
>>> On Apr 19, 2012, at 1:35 PM, Avi Marcus wrote:
>>> 
>>> This sounds like a totally different issue.
>>> Please try using FreeSWITCH's encoding functionality:
>>> 
>>> http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding
>>> 
>>> -Avi
>>> 
>>> 
>>> On Thu, Apr 19, 2012 at 8:29 PM, Bernard Fluixa <fluixab at bellsouth.net>
>>> wrote:
>>>> 
>>>> There's still something I'm not clear with. Here's what I did
>>>> 
>>>> 1) Converted a ".wav" file to Speex with speex : speexenc a.wav a.SPEEX
>>>> 2) Call a user with x-Lite soft phone: originate user/1001
>>>> &playback(/home/sounds/a.SPEEX)
>>>> 
>>>> ==> file is open and played but one can hear only noise squelch. What am
>>>> I doing wrong?
>>>> 
>>>> FYI, I then converted a.SPEEX file back to wav format and could play that
>>>> wav file.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Apr 19, 2012, at 1:19 PM, Anthony Minessale wrote:
>>>> 
>>>>> no you, cant modify it.  This is a function of mod_native_file and its
>>>>> limited to matching the iananames they registered the codec with.
>>>>> 
>>>>> 
>>>>> 
>>>>> On Thu, Apr 19, 2012 at 11:59 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>> It seems all the codecs are are chosen in uppercase, as it seems to be
>>>>>> listed here: http://wiki.freeswitch.org/wiki/Mod_native_file
>>>>>> The name for SPEEX codec is SPEEX, not spx, so that's why it chose it.
>>>>>> You can probably modify mod_speex to add an alias to load .spx files
>>>>>> also.
>>>>>> 
>>>>>> -Avi
>>>>>> 
>>>>>> 
>>>>>> On Thu, Apr 19, 2012 at 7:25 PM, Bernard Fluixa
>>>>>> <fluixab at bellsouth.net>
>>>>>> wrote:
>>>>>>> 
>>>>>>> Avi - WeIl, I recorded one file yesterday with ".SPEEX" extension
>>>>>>> FROM
>>>>>>> Freeswitch and could play it with playback command. No problem.
>>>>>>> However, it
>>>>>>> is my understanding that speex format files have ".spx" extension.
>>>>>>> Why
>>>>>>> doesn't Freeswitch allows for ".spx" extension?
>>>>>>> 
>>>>>>> .speex extension are not accepted.
>>>>>>> 
>>>>>>> Thanks again
>>>>>>> 
>>>>>>> Bernard
>>>>>>> 
>>>>>>> On Apr 19, 2012, at 11:50 AM, Avi Marcus wrote:
>>>>>>> 
>>>>>>> Didn't we answer this yesterday? Bernard, have you tried it yet with
>>>>>>> a
>>>>>>> .SPEEX (or .speex?) extension?
>>>>>>> 
>>>>>>> -Avi
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Thu, Apr 19, 2012 at 6:38 PM, Bernard Fluixa
>>>>>>> <fluixab at bellsouth.net>
>>>>>>> wrote:
>>>>>>>> 
>>>>>>>> Hello,
>>>>>>>> 
>>>>>>>> I cannot play ".spx" (speex) file in commands such as originate?
>>>>>>>> &playback(file.spx). I double-checked that mod_speex is loaded OK.
>>>>>>>> Freeswitch returns
>>>>>>>> 
>>>>>>>> 2012-04-19 11:35:40.532068 [ERR] switch_core_file.c:122 Invalid file
>>>>>>>> format [spx] for [test.spx]!
>>>>>>>> 
>>>>>>>> I converted file to wav format and could play it (speexdec test.spx
>>>>>>>> test.wav) with playback command. I probably missed a point here but
>>>>>>>> I would
>>>>>>>> like to know how I can play speex files without having to convert
>>>>>>>> them to
>>>>>>>> wav first.
>>>>>>>> 
>>>>>>>> Anyone?
>>>>>>>> 
>>>>>>>> Thank you
>>>>>>>> 
>>>>>>>> Bernard
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> 
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> --
>>>>> Anthony Minessale II
>>>>> 
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>> 
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>> 
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:+19193869900
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> 
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> 
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>> 
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Kristian Kielhofner
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Rick at Claritycomputer.com  Tue Apr 24 03:29:12 2012
From: Rick at Claritycomputer.com (Rick Guyton)
Date: Mon, 23 Apr 2012 23:29:12 +0000
Subject: [Freeswitch-users] BLF in Multi-tenant config
Message-ID: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>

We have 3 different companies in this office and we all share the same freeswitch server, I'll call then CompanyA, CompanyB and CompanyC. CompanyA, B and C all employ their own receptionists, and they are all assigned extension 101 on in their respective domains. However, the receptionists all work very tightly together and basically share work between them. So, I need to assign the BLF status of 101 at CompanyA, 101 at CompanyB and 101 at CompanyC to indicator lights across all phones in the building. My problem is that users in CompanyA can't see the status of 101 at CompanyB or 101 at CompanyC. Users in CompanyB can't see 101 at CompanyA or 101 at CompanyC and so on...

So, my question is, how do I allow users of one domain in a multi-tenant configuration to see the status of other users in a different domain?
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From bdfoster at endigotech.com  Tue Apr 24 05:21:22 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Mon, 23 Apr 2012 21:21:22 -0400
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
Message-ID: <CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>

I'm not sure that multi-tenant wouldn't support that. I think the point of
multi-tenant is to keep everything seperated. There might be a hack for
this, though. Nothing comes to mind for me.

-BDF
On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com> wrote:

>  We have 3 different companies in this office and we all share the same
> freeswitch server, I?ll call then CompanyA, CompanyB and CompanyC.
> CompanyA, B and C all employ their own receptionists, and they are all
> assigned extension 101 on in their respective domains. However, the
> receptionists all work very tightly together and basically share work
> between them. So, I need to assign the BLF status of 101 at CompanyA,
> 101 at CompanyB and 101 at CompanyC to indicator lights across all phones in
> the building. My problem is that users in CompanyA can?t see the status of
> 101 at CompanyB or 101 at CompanyC. Users in CompanyB can?t see 101 at CompanyA or
> 101 at CompanyC and so on?****
>
> ** **
>
> So, my question is, how do I allow users of one domain in a multi-tenant
> configuration to see the status of other users in a different domain?****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mitch.capper at gmail.com  Tue Apr 24 05:23:32 2012
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 23 Apr 2012 18:23:32 -0700
Subject: [Freeswitch-users] client change IP address while ringing
In-Reply-To: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>
References: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>
Message-ID: <CALn233qM80E8izqKwkyM9sqEg9sJ_2Np-HoJ8FHbZN6683TYKQ@mail.gmail.com>

Try changing the mobile client to register via TCP I would assume then
the IP will not change as that would break the TCP connection:)

~Mitch


From chris.chen2004 at gmail.com  Tue Apr 24 05:29:32 2012
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 23 Apr 2012 21:29:32 -0400
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
Message-ID: <CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>

Agree with BDF.

By design, in a multi-tenant environment, 101 at companyA should only have BLF
status for any extension within companyA,  otherwise why should you have
multi-tenant setup, just have them all in the same domain. We should really
appreciate Tony crew having built the FS to be really capable of providing
multi-tenant functions without too many tweaks.

Just my two cents.

Chris


On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> I'm not sure that multi-tenant wouldn't support that. I think the point of
> multi-tenant is to keep everything seperated. There might be a hack for
> this, though. Nothing comes to mind for me.
>
> -BDF
> On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com> wrote:
>
>>  We have 3 different companies in this office and we all share the same
>> freeswitch server, I?ll call then CompanyA, CompanyB and CompanyC.
>> CompanyA, B and C all employ their own receptionists, and they are all
>> assigned extension 101 on in their respective domains. However, the
>> receptionists all work very tightly together and basically share work
>> between them. So, I need to assign the BLF status of 101 at CompanyA,
>> 101 at CompanyB and 101 at CompanyC to indicator lights across all phones in
>> the building. My problem is that users in CompanyA can?t see the status of
>> 101 at CompanyB or 101 at CompanyC. Users in CompanyB can?t see 101 at CompanyAor 101 at CompanyCand so on?
>> ****
>>
>> ** **
>>
>> So, my question is, how do I allow users of one domain in a multi-tenant
>> configuration to see the status of other users in a different domain?****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ian.mcmaster at gmail.com  Tue Apr 24 06:12:13 2012
From: ian.mcmaster at gmail.com (Ian McMaster)
Date: Mon, 23 Apr 2012 22:12:13 -0400
Subject: [Freeswitch-users] Park and Page use case
Message-ID: <CAL0C-v89p6E04v3kpJurk6WgRiQo-nN=PWWCz4EkPK56y-p66g@mail.gmail.com>

As this is my first question, please let me first praise the FreeSWITCH
team for a fantastic product and all the efforts of the community.  The two
books really helped convert all my Asterisk AGI / AMI java code to
FreeSWITCH.  All IVR play/record/dtmf, etc is working perfectly with
FreeSwitch now (and the IVR application is quite large).
I came to FreeSwitch after limitations/frustrations with Asterisk.  Happy
to be aboard, and hope to contribute in time.

Use case:  Controlling an IVR caller using the event socket; caller chooses
an option which will park the caller, and page out to a group of phones for
someone to pick up the parked caller.  If the park is answered the script
exits, or if the park times out then an alternate action is taken.

The valet_park command is working fine, and I can decipher through the
ChannelExecuteComplete event whether the park was retrieved or not.

I am struggling a bit with the paging to multiple polycom phones.

- using something like 'originate
{sip_auto_answer=true}user/1001,user/1002,...
 &playback(announce_page.PCMU) has limited success - my understanding is
that the first phone to answer 'wins' and the other phones do not receive
the playback.
- I have better success with conference
(a)  Invite phones into the conference: conference page_group dial
{sip_auto_answer=true}user/1001 ... etc
(b)  Play file:  conference page_group play announce_page.PCMU (although I
am having a codec problem here!)
(c)  End the conference:  conference page_group hup all
While the conference by nature has certain pre-stocked files (i.e. you are
the first caller, and music on hold, etc), I believe I can silence them all
with a custom conference.

My only show stopper is that if a page goes out to a phone with two line
buttons, and is already in use (on one line), then things go bad.  The
incoming conference request is knocking the existing call on hold, the page
is playing out, and the user must resume the call with a soft button.

Questions:
1)  Is there a better approach to paging multiple phones that I have yet to
try?
2)  Is there a way to avoid paging a phone that already has a caller on it?
3)  Is there conference variable to set the codec to allow the PCMU file to
play cleanly (it does play cleanly using originate without setting any
explicit codec).

Thank you in advance,
Ian.
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From Rick at Claritycomputer.com  Tue Apr 24 08:43:45 2012
From: Rick at Claritycomputer.com (Rick Guyton)
Date: Tue, 24 Apr 2012 04:43:45 +0000
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
	<CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>
Message-ID: <B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>

To be clear, I'm in utter amazement of what freeswitch can do and appreciate everyone who has contributed to making such an awesome application. So, when I can't do something, I assume it is because of my own stupidity. I had assumed that I needed to make a "permissions" change somewhere or add the users from other domains as an alias or some sort. If it can't be done, that makes sense. I meant no disrespect to Tony or anyone else who has contributed. I would just put them all in the same domain, but I must separate the companies for mostly political reasons.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Chen
Sent: Monday, April 23, 2012 6:30 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] BLF in Multi-tenant config

Agree with BDF.

By design, in a multi-tenant environment, 101 at companyA should only have BLF status for any extension within companyA,  otherwise why should you have multi-tenant setup, just have them all in the same domain. We should really appreciate Tony crew having built the FS to be really capable of providing multi-tenant functions without too many tweaks.

Just my two cents.

Chris

On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:

I'm not sure that multi-tenant wouldn't support that. I think the point of multi-tenant is to keep everything seperated. There might be a hack for this, though. Nothing comes to mind for me.

-BDF
On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com<mailto:Rick at claritycomputer.com>> wrote:
We have 3 different companies in this office and we all share the same freeswitch server, I'll call then CompanyA, CompanyB and CompanyC. CompanyA, B and C all employ their own receptionists, and they are all assigned extension 101 on in their respective domains. However, the receptionists all work very tightly together and basically share work between them. So, I need to assign the BLF status of 101 at CompanyA, 101 at CompanyB and 101 at CompanyC to indicator lights across all phones in the building. My problem is that users in CompanyA can't see the status of 101 at CompanyB or 101 at CompanyC. Users in CompanyB can't see 101 at CompanyA or 101 at CompanyC and so on...

So, my question is, how do I allow users of one domain in a multi-tenant configuration to see the status of other users in a different domain?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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________________________________
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From bdfoster at endigotech.com  Tue Apr 24 09:04:39 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 24 Apr 2012 01:04:39 -0400
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
	<CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>
	<B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>
Message-ID: <CAKV-jFydnHBYZkrPVxS8Py6yRijThc6NF_f54mEVpAUHhuEuSw@mail.gmail.com>

On Tue, Apr 24, 2012 at 12:43 AM, Rick Guyton <Rick at claritycomputer.com>wrote:

>  To be clear, I?m in utter amazement of what freeswitch can do and
> appreciate everyone who has contributed to making such an awesome
> application.
>

Welcome to the club ;-)


> So, when I can?t do something, I assume it is because of my own stupidity.
> I had assumed that I needed to make a ?permissions? change somewhere or add
> the users from other domains as an alias or some sort. If it can?t be done,
> that makes sense. I meant no disrespect to Tony or anyone else who has
> contributed. I would just put them all in the same domain, but I must
> separate the companies for mostly political reasons.
>

Understandable.

Depending on your setup, you might just be able to do this on the phones
themselves. I don't know if you use these, but you know the little
attendant consoles you can plug into phones like polycom and aastra? Can
you register to the lines you want to monitor with those? I've never
actually had one (I'd like to get one for testing) but I'm fairly certain
you can. That would also depend on your network setup. If you've got all
the companies on their own LAN's then you're screwed unless freeswitch can
be fully accessed on each of those LAN's. Might be something to check out.

> * *
>
-BDF

> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Chen
> *Sent:* Monday, April 23, 2012 6:30 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] BLF in Multi-tenant config****
>
> ** **
>
> Agree with BDF.****
>
> ** **
>
> By design, in a multi-tenant environment, 101 at companyA should only have
> BLF status for any extension within companyA,  otherwise why should you
> have multi-tenant setup, just have them all in the same domain. We should
> really appreciate Tony crew having built the FS to be really capable of
> providing multi-tenant functions without too many tweaks. ****
>
> ** **
>
> Just my two cents.****
>
> ** **
>
> Chris****
>
> ** **
>
> On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster <bdfoster at endigotech.com>
> wrote:****
>
> I'm not sure that multi-tenant wouldn't support that. I think the point of
> multi-tenant is to keep everything seperated. There might be a hack for
> this, though. Nothing comes to mind for me.****
>
> -BDF****
>
> On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com> wrote:**
> **
>
>   We have 3 different companies in this office and we all share the same
> freeswitch server, I?ll call then CompanyA, CompanyB and CompanyC.
> CompanyA, B and C all employ their own receptionists, and they are all
> assigned extension 101 on in their respective domains. However, the
> receptionists all work very tightly together and basically share work
> between them. So, I need to assign the BLF status of 101 at CompanyA,
> 101 at CompanyB and 101 at CompanyC to indicator lights across all phones in
> the building. My problem is that users in CompanyA can?t see the status of
> 101 at CompanyB or 101 at CompanyC. Users in CompanyB can?t see 101 at CompanyA or
> 101 at CompanyC and so on?****
>
>  ****
>
> So, my question is, how do I allow users of one domain in a multi-tenant
> configuration to see the status of other users in a different domain?****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
> ** **
>  ------------------------------
>
> If this email is spam, report it to www.OnlyMyEmail.com<https://support.onlymyemail.com/view/report_spam/MTM3Mjc1OjE0MTI0MDQ4MjI6cmlja0BjbGFyaXR5Y29tcHV0ZXIuY29tOmRlbGl2ZXJlZA>
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
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corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
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From oh2aun at gmail.com  Tue Apr 24 10:01:35 2012
From: oh2aun at gmail.com (Mikael Nordman)
Date: Tue, 24 Apr 2012 09:01:35 +0300
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAD-qRkgq=c0-LUCYHZWTu9pGzLFqx3VJnrp5F3F1REvDbvaJvw@mail.gmail.com>
References: <CAKyTn5SomTJ_nV1VdvG7ivZ5sicXB+eUk6BUhJ1zGqqE2Y7sPA@mail.gmail.com>
	<CAD-qRkgq=c0-LUCYHZWTu9pGzLFqx3VJnrp5F3F1REvDbvaJvw@mail.gmail.com>
Message-ID: <CAKyTn5Tkr7tYDaZZyYcXWOnr43-i4Sp7_hJtG+3PhrZV4hzBpw@mail.gmail.com>

Thanks Chris,

It works like a clock now.

Mike

On Mon, Apr 23, 2012 at 4:33 PM, Christopher Rienzo <cmrienzo at gmail.com>wrote:

> I have no idea.  Use silence_stream://  if you don't want anything to be
> played. Set the silence to be the same length as the speech timeout.  For
> example: silence_stream://5000
>
> If you observe incorrect behavior while playing silence, then open a jira
> ticket so we can track it.
>
> Chris
>
>
>
> On Mon, Apr 23, 2012 at 9:20 AM, Mikael Nordman <oh2aun at gmail.com> wrote:
>
>> Hi,
>> I have a asr dialogue with two consecutive play_and_detect_speech calls.
>>
>> The say: part is empty and the detect: part uses nuance grxml grammar.
>>
>> In the second play_and_detect invocation freeswitch looses some of the
>> userplane data,
>> thus causing the speech detection to fail.
>>
>> From the userplane data forwarded by freeswitch to the Nuance NSS server,
>> about one second of speech is missing in the beginning of the phrase
>>
>> If I add say: with some content, then all works fine.
>>
>> What could cause this loss of userplane data ?
>>
>> Cheers /Mike
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario at ims.bg  Tue Apr 24 10:19:23 2012
From: mario at ims.bg (Mario Karakanovski)
Date: Tue, 24 Apr 2012 09:19:23 +0300
Subject: [Freeswitch-users] client change IP address while ringing
In-Reply-To: <CALn233qM80E8izqKwkyM9sqEg9sJ_2Np-HoJ8FHbZN6683TYKQ@mail.gmail.com>
References: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>
	<CALn233qM80E8izqKwkyM9sqEg9sJ_2Np-HoJ8FHbZN6683TYKQ@mail.gmail.com>
Message-ID: <BCCA7469823E440096A6D6BC76E07037@imsdeveloper>

Thank you Mitch,

	The client use TCP, but the problem is that it must change IP
address, because it's switch between WiFi and 3G. So, there is no chance to
force it using the same IP address.

Regards,
	Mario

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
Capper
Sent: Tuesday, April 24, 2012 4:24 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] client change IP address while ringing

Try changing the mobile client to register via TCP I would assume then
the IP will not change as that would break the TCP connection:)

~Mitch

Hi guys,

 

            I have an interesting situation.

I have two mobile clients and connect them using bridge. It is working
without problems, except when the called party changes its IP address
between ringing and answer.

Does anybody deal with such situation?

 

Regards,

            Mario  
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From bdfoster at endigotech.com  Tue Apr 24 10:23:21 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Tue, 24 Apr 2012 02:23:21 -0400
Subject: [Freeswitch-users] mod_lumenvox ?
In-Reply-To: <CAKV-jFw0zZUZ75S1h01f92CbzNarA8uGNk7FyU6iUf5shWCA5w@mail.gmail.com>
References: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>
	<CAKV-jFw0zZUZ75S1h01f92CbzNarA8uGNk7FyU6iUf5shWCA5w@mail.gmail.com>
Message-ID: <CAKV-jFwPUdWD+p0nP9-DriveGvJWmieMSq5VvDFgTacQuaCg4A@mail.gmail.com>

A little searching turned up that mod_lumenvox is not and will not be
offered. Looks like the company who owns that technology wanted some $$
from the guy who developed the module, so it was taken out of the GIT repo.

-BDF

On Sat, Apr 21, 2012 at 12:35 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Same place mod_ldap went? No clue.
>
> -BDF
>
> On Fri, Apr 20, 2012 at 11:56 PM, Anita Hall <anita.hall at simmortel.com>wrote:
>
>> Whatever happened to mod_lumenvox ?
>>
>> regards,
>> Anita
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.
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From anita.hall at simmortel.com  Tue Apr 24 10:42:04 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Tue, 24 Apr 2012 12:12:04 +0530
Subject: [Freeswitch-users] mod_lumenvox ?
In-Reply-To: <CAKV-jFwPUdWD+p0nP9-DriveGvJWmieMSq5VvDFgTacQuaCg4A@mail.gmail.com>
References: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>
	<CAKV-jFw0zZUZ75S1h01f92CbzNarA8uGNk7FyU6iUf5shWCA5w@mail.gmail.com>
	<CAKV-jFwPUdWD+p0nP9-DriveGvJWmieMSq5VvDFgTacQuaCg4A@mail.gmail.com>
Message-ID: <CAGKPtBuh-S-RTbFnkVAMCgtpdwqRXANOg955Sw_iRCx0-V1xLQ@mail.gmail.com>

I hope they are getting enough $$ now from those who will not be using
their engine and try out others instead :D

regards,
Anita



On Tue, Apr 24, 2012 at 11:53 AM, Brian Foster <bdfoster at endigotech.com>wrote:

> A little searching turned up that mod_lumenvox is not and will not be
> offered. Looks like the company who owns that technology wanted some $$
> from the guy who developed the module, so it was taken out of the GIT repo.
>
> -BDF
>
>
> On Sat, Apr 21, 2012 at 12:35 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Same place mod_ldap went? No clue.
>>
>> -BDF
>>
>> On Fri, Apr 20, 2012 at 11:56 PM, Anita Hall <anita.hall at simmortel.com>wrote:
>>
>>> Whatever happened to mod_lumenvox ?
>>>
>>> regards,
>>> Anita
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Brian D. Foster
>> Endigo Computer LLC
>> Email: bdfoster at endigotech.com
>> Phone: 317-800-7876
>> Indianapolis, Indiana, USA
>>
>> This message contains confidential information and is intended for those
>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
>> you are not the intended recipient you are notified that disclosing,
>> copying, distributing or taking any action in reliance on the contents of
>> this information is strictly prohibited. E-mail transmission cannot be
>> guaranteed to be secure or error-free as information could be intercepted,
>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
>> The sender therefore does not accept liability for any errors or omissions
>> in the contents of this message, which arise as a result of e-mail
>> transmission. If verification is required please request a hard-copy
>> version.
>>
>>
>
>
> --
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for those
> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
> you are not the intended recipient you are notified that disclosing,
> copying, distributing or taking any action in reliance on the contents of
> this information is strictly prohibited. E-mail transmission cannot be
> guaranteed to be secure or error-free as information could be intercepted,
> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
> The sender therefore does not accept liability for any errors or omissions
> in the contents of this message, which arise as a result of e-mail
> transmission. If verification is required please request a hard-copy
> version.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From tomp at tomp.co.uk  Tue Apr 24 12:10:46 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Tue, 24 Apr 2012 09:10:46 +0100
Subject: [Freeswitch-users] mod_httapi
In-Reply-To: <4F95819C.7090502@tomp.co.uk>
References: <4F9435D8.90307@tomp.co.uk> <4F944515.609@tomp.co.uk>
	<4F95819C.7090502@tomp.co.uk>
Message-ID: <fcba74f73079a74f45d882913aea0194.squirrel@my.netvps.co.uk>

Antony, that is awesome. I was hoping it worked like that.

That means that my app can make dynamic decisions (like time routing with
with timezone support) by it requesting for more work at the end of the
initial dialplan.

Cheers
Tom

> Hi Kyle,
>
> Thanks for your response. I wasn't being clear though.
>
> I understand that
>
> <action application="httapi"
> data="http://192.168.100.197/cgi-bin/test.cgi"/>  in the dialplan will
> cause Freeswitch to retrieve the XML to perform work in the dialplan.
>
> However in the wiki http://wiki.freeswitch.org/wiki/Mod_httapi
>   under the "Work Tag" section, each type of work describes two
> attributes:
>
> action and temp-action, described as "change the new target url" and
> "change url to submit to just for the next loop".
>
> What is the "loop" it is describing here, and is this a kind of 'callback'
> to the test.cgi allowing the application to keep control of the dialplan
> after it has sent the initial response?
>
> Thanks again
> Tom
>
>
> On 22/04/12 18:51, Tom Parrott wrote:
>> Thanks Michael, yes will certainly help keep the page updated if any
>> more info comes to light.
>>
>> Thanks Avi, when you say "future httpapi requests" do you think that
>> is when the httapi application is executed again, or is it a type of
>> 'callback' so that when that step in the dialplan is finished the HTTP
>> API being called can instruct Freeswitch to perform another step in
>> the dialplan.
>>
>> Tom
>>
>> On 22/04/12 17:46, Tom Parrott wrote:
>>> Hi,
>>>
>>> Does anyone have a short guide or blog post on using the mod_httapi.
>>>
>>> The wiki page is a bit sparse on details of how to actually use this
>>> new module, and have not been able to make much progress with it.
>>>
>>> One of my main questions is what is the "action" property of the work
>>> element for.
>>>
>>> The wiki says it "Change the new target url."
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_httapi
>>>
>>> But what does that mean?
>>>
>>> Thanks
>>> Tom
>




From yehavi.bourvine at gmail.com  Tue Apr 24 12:40:38 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 24 Apr 2012 11:40:38 +0300
Subject: [Freeswitch-users] Incoming fax stopped working when removing
	mod_fax...
Message-ID: <CAAtWKeGF4pbyR+bnu2oMx1-kmBHrOpM1xYnX4uhObkmUqz4SaQ@mail.gmail.com>

Hi,

  I am trying to upgrade from a half year olf FS version to the latest one
and face some problems with fax. The fax arrives from the PSTN through an
Audiocodes MSBG-1,000 gateway.

With the old version, when I have mod_fax loaded, FS detects that this is a
fax call, sends a re-INVITE to the MSBG and enters T.38 mode; this works ok.
With the old version, when mod_fax is *unloaded*, FS enters T.38 mode *without
*sending re-INVITE, thus the fax fails.

The latest FS version (where mod_fax is not available anymore) works as the
second situation above and fax fails. I guess I am missing something in the
configs, but I can't find what...

I have in my dialplan:

     <action application="set" data="fax_enable_t38_request=true"/>
      <action application="set" data="fax_enable_t38=true"/>
and I've also tried to add:
    <action application="set" data="execute_on_answer_6=t38_gateway self"/>
but nothing helps. Any idea what I am doing wrong?

                              Thanks! __Yehavi:
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From peter.olsson at visionutveckling.se  Tue Apr 24 12:51:50 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 24 Apr 2012 08:51:50 +0000
Subject: [Freeswitch-users] Incoming fax stopped working when
	removing	mod_fax...
Message-ID: <1FFF97C269757C458224B7C895F35F15097CF4@cantor.std.visionutv.se>

Have you loaded mod_spandsp? This is the replacement for mod_fax.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yehavi Bourvine
Skickat: den 24 april 2012 10:41
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Incoming fax stopped working when removing mod_fax...

Hi,

  I am trying to upgrade from a half year olf FS version to the latest one and face some problems with fax. The fax arrives from the PSTN through an Audiocodes MSBG-1,000 gateway.

With the old version, when I have mod_fax loaded, FS detects that this is a fax call, sends a re-INVITE to the MSBG and enters T.38 mode; this works ok.
With the old version, when mod_fax is unloaded, FS enters T.38 mode without sending re-INVITE, thus the fax fails.

The latest FS version (where mod_fax is not available anymore) works as the second situation above and fax fails. I guess I am missing something in the configs, but I can't find what...

I have in my dialplan:

     <action application="set" data="fax_enable_t38_request=true"/>
      <action application="set" data="fax_enable_t38=true"/>
and I've also tried to add:
    <action application="set" data="execute_on_answer_6=t38_gateway self"/>
but nothing helps. Any idea what I am doing wrong?

                              Thanks! __Yehavi:
!DSPAM:4f96658d32762460221281!
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From yehavi.bourvine at gmail.com  Tue Apr 24 13:12:06 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 24 Apr 2012 12:12:06 +0300
Subject: [Freeswitch-users] Incoming fax stopped working when removing
	mod_fax...
In-Reply-To: <1FFF97C269757C458224B7C895F35F15097CF4@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15097CF4@cantor.std.visionutv.se>
Message-ID: <CAAtWKeFSNP-=WhVVCwxFBwhUdUOVYLL6xCNncmZ5fbMqeZf3Pw@mail.gmail.com>

Yes, it is loaded. I guess it won't send T.38 signals without this module
being loaded.

Maybe I missed something in my description: FS starts sendnig T.38 data,
but it does not sends re-INVITE to the other side, so the other side (MSBG)
keeps sending audio instead of entering T.38 mode.

                  Thanks, __Yehavi:
2012/4/24 Peter Olsson <peter.olsson at visionutveckling.se>

>  Have you loaded mod_spandsp? This is the replacement for mod_fax.****
>
> ** **
>
> /Peter****
>
> ** **
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Yehavi Bourvine
> *Skickat:* den 24 april 2012 10:41
> *Till:* FreeSWITCH Users Help
> *?mne:* [Freeswitch-users] Incoming fax stopped working when removing
> mod_fax...****
>
> ** **
>
> Hi,****
>
>  ****
>
>   I am trying to upgrade from a half year olf FS version to the latest one
> and face some problems with fax. The fax arrives from the PSTN through an
> Audiocodes MSBG-1,000 gateway.****
>
>  ****
>
> With the old version, when I have mod_fax loaded, FS detects that this is
> a fax call, sends a re-INVITE to the MSBG and enters T.38 mode; this works
> ok.****
>
> With the old version, when mod_fax is *unloaded*, FS enters T.38 mode *without
> *sending re-INVITE, thus the fax fails.****
>
>  ****
>
> The latest FS version (where mod_fax is not available anymore) works as
> the second situation above and fax fails. I guess I am missing something in
> the configs, but I can't find what...****
>
>  ****
>
> I have in my dialplan:****
>
>  ****
>
>      <action application="set" data="fax_enable_t38_request=true"/>
>       <action application="set" data="fax_enable_t38=true"/>****
>
> and I've also tried to add:****
>
>     <action application="set" data="execute_on_answer_6=t38_gateway self"/>
> ****
>
> but nothing helps. Any idea what I am doing wrong?****
>
>  ****
>
>                               Thanks! __Yehavi:****
>
> !DSPAM:4f96658d32762460221281! ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From chrisbware at yahoo.it  Tue Apr 24 14:12:43 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Tue, 24 Apr 2012 11:12:43 +0100 (BST)
Subject: [Freeswitch-users] Freeswitch speaks italian
Message-ID: <1335262363.89683.YahooMailNeo@web132305.mail.ird.yahoo.com>

Hi,

some time ago, Mr. Marco Menardi and his wife Paola (the voice) made a set of audio files for Asterisk in italian, under creative commons license.
Using Audacity I rearranged them to be used with FS. ?Translated folders are:

ascii ?
currency ?
digits ?
phonetic-ascii ?
time ?
voicemail


Only 8Khz version, sorry. ?Link to my blog, where you can download files:?http://chrisbware.wordpress.com/2012/04/12/freeswitch-parla-italiano/
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From anton.jugatsu at gmail.com  Tue Apr 24 14:36:19 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Tue, 24 Apr 2012 14:36:19 +0400
Subject: [Freeswitch-users] Freeswitch speaks italian
In-Reply-To: <1335262363.89683.YahooMailNeo@web132305.mail.ird.yahoo.com>
References: <1335262363.89683.YahooMailNeo@web132305.mail.ird.yahoo.com>
Message-ID: <CANBhyPDrJog_m0gbkLGLNQr0YBjJBkwsKBRERfga2b1GtKY_Xg@mail.gmail.com>

Congratulations!

24 ?????? 2012 ?. 14:12 ???????????? Chris B. Ware <chrisbware at yahoo.it>???????:

> Hi,
>
> some time ago, Mr. Marco Menardi and his wife Paola (the voice) made a set
> of audio files for Asterisk in italian, under creative commons license.
> Using Audacity I rearranged them to be used with FS.  Translated folders
> are:
>
> ascii
> currency
> digits
> phonetic-ascii
> time
> voicemail
>
> Only 8Khz version, sorry.  Link to my blog, where you can download files:
> http://chrisbware.wordpress.com/2012/04/12/freeswitch-parla-italiano/
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From dinhthuc1303 at gmail.com  Tue Apr 24 15:14:03 2012
From: dinhthuc1303 at gmail.com (dinh thuc)
Date: Tue, 24 Apr 2012 18:14:03 +0700
Subject: [Freeswitch-users] detect tone
Message-ID: <CAKMOEZmqJaQDKZhu=KEL5_1vjhA9yuvm4he=8UECm-9gtSDuDg@mail.gmail.com>

Hi

I have a question about detect tone:

I have 2 freeswitch server and I use mod_spandsp to detect tone:

1. I use SIP Phone call to mod_spandsp, it  make call to the second
freeswitch.
2. Second freeswitch will answer and  genarate tone like  <action
application="playback"
data="tone_stream://%%(250,250,480,620);%(500,500,480,620)"/>
3. mod_spandsp will start_detect_tone

I have a problem:

 I create make call from mod_spandsp to second freeswitch I will have 2
session ( channels)

1st session: SIP Phone call to mod_spandsp
2st session: mod_spandsp call to the second freeswitch


I try to dectect 1st session, then I can dectect tone

But I try to dectect 2st session, then I see that dectect tone don't run


What is the problem with 2st session ? Whycan't I  detect tone ?

And I also problem is I can't detect all tone that I send from second
freeswitch and It's too difficult to detect tone


B&R

Dinh Thuc
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From miha at softnet.si  Tue Apr 24 15:56:10 2012
From: miha at softnet.si (Miha)
Date: Tue, 24 Apr 2012 13:56:10 +0200
Subject: [Freeswitch-users] Freeswitch and Opensips
In-Reply-To: <4F911AFE.2@softnet.si>
References: <4F8D3AB1.1030207@softnet.si>
	<CANBhyPBfXzDo3f0x7VOUw+9nc0sKrGxQSMwc_LJRCC-7b1MumA@mail.gmail.com>
	<4F911AFE.2@softnet.si>
Message-ID: <4F9694DA.9070104@softnet.si>

Hi,

I have configured Opensips to behave also as registrar. Only invites are 
load_balance to one of the FS boxes. What must I do in FS configuration 
as FS now reply  Originate Failed.  Cause: USER_NOT_REGISTERED.
This is logically to me as users are registering to Opensips and FS 
handles media and etc.

Is there any threat on wiki about this (I did not find)?

p.s.: I have put opensips IP to FS ACL.

Thanks for all your help!

Regards,
MIha

On 4/20/2012 10:14 AM, Miha wrote:
> On 4/17/2012 2:15 PM, Anton Kvashenkin wrote:
>> Good read 
>> http://blog.2600hz.com/post/18096749028/understanding-how-fs-supports-opensips-as-a 
>>
>>
>> 17 ?????? 2012 ?. 13:41 ???????????? Miha <miha at softnet.si 
>> <mailto:miha at softnet.si>> ???????:
>>
>>     Hi to all,
>>
>>     I have configured opensips that works like load_balancer (wiki on
>>     FS page).
>>     So, we have SBC, trunk is made to Opensips, than opensips is
>>     load_balacing to FSs.
>>     When you call from FS, call is going to Opensips, then opensips
>>     redirect the call to SBC, if call must go outside.
>>
>>     My question:
>>
>>     FSs boxes are all the same (same dialplan, directory
>>     configuration, etc.). What happens if call is made to a user,
>>     which is registered on one of the FS box? I guess I must change
>>     dialplan and set that all trafic FS send to opensips (not
>>     internally), because if someone calls to a local user, but user
>>     is registered on other FS box, FS will behave as user is not
>>     registered.
>>
>>     So, will opensips know which user is registered on which FS box
>>     when opensips will be doing load_balacing?
>>
>>     Thanks for explenation.
>>
>>     Regards,
>>     Miha
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> HI,
>
> thank you all for your answers . So If I would like to do 
> load_balacing with opensips I would need some kind of logic for every 
> registered phone I must know to which FS is registerd, so I would know 
> to which one I must FDW call.
>
> But in this case this script do not work (as is written on FS wiki):
>
>   if (is_method("INVITE")) {
>                  if (!load_balance("1","pstn","1")) {
>                          send_reply("503","Service Unavailable");
>                          exit;
>                  }
>          }
>          else if (is_method("REGISTER")) {
>                  if (!ds_select_dst("1", "0")) {
>                          send_reply("503","Service Unavailable");
>                          exit;
>                  }
>          }
>          else {
>                  send_reply("405","Method Not Allowed");
>                  exit;
>          }
>
>          # route the request
>          if (!t_relay()) {
>                  sl_reply_error();
>          }
>
>
>
>
> Example:. You have three users.A and B user are registered to FS1 and C user to FS2. A and C are talking if B gets a call as Opensips do not know to which box B is registered
> It could FWD it to FS2, which will not work as B is registered to FS1. 
> Am I right?
>
> Rergards,
> Miha
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From koralu at gmail.com  Tue Apr 24 14:19:12 2012
From: koralu at gmail.com (Cucu Florian)
Date: Tue, 24 Apr 2012 13:19:12 +0300
Subject: [Freeswitch-users] Bypass Media not work
Message-ID: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>

Hello,

I tried to test the behavior of "bypass media" in FS with the following
scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)

On each Xlite I set one single codec (that is currently unsupported by FS)
as it said in wiki page: *"This is useful if you have two end-points that
need to use a codec that is currently not supported in FreeSWITCH "*

As the FS tutorial said *"Before executing the bridge action you must set
the "bypass_media" flag to true. "* so I put the following line in my dial
plan.

<extension name="extension-intercom">
    <condition field="destination_number" expression="1001">
       * <action application="set" data="bypass_media=true"/>*
        <action application="bridge" data="user/1001"/>
    </condition>
</extension>


<extension name="extension-intercom">
    <condition field="destination_number" expression="1000">
        *<action application="set" data="bypass_media=true"/>*
        <action application="bridge" data="user/1000"/>
    </condition>
</extension>

When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
ok. But when I put only codecs that FS do not support(ex.
iLBC,BroadVoice32) the call failed. What am I missing?

I made this scenario because I need to route via FS some calls with G729r8
codec and I need no codec negotiation.

Ty
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From haythem.horbit at gmail.com  Tue Apr 24 15:46:32 2012
From: haythem.horbit at gmail.com (Haythem Horbit)
Date: Tue, 24 Apr 2012 13:46:32 +0200
Subject: [Freeswitch-users] How to enable Hardware DTMF detection
	(Dahdi/Digium card)
Message-ID: <CANmn8agD4c8ZgRTmekjqsqFVMOhj5Ep1yeh15vMb5aMWBDKW7g@mail.gmail.com>

Hi,

We experience some issues to capture SIA ContactID messages from an alarm
system, with Freeswitch,
the problem is that some DTMFs are missed randomly, so that we can't have a
correct message
we use a Digium card 'AEX2400' with DAHDI drivers, the alarm system is
connected to an FXO port.

We see in the freeswitch log that software DTMF detection is enabled, may
be the issue comes from this fact...

So our question is: how can we enable hardware DTMF detection in DADHI
configuration with freeswitch,
and how to disable software DTMF detection in freeswitch

-- 

*Haythem HORBIT
Responsable d?veloppement
NGI Maghreb*
53, rue 8601 - 2035 Charguia I
Tunis - TUNISIE

T?l.    : +216 71 808 565
Fax    : +216 71 808 395
Gsm   : +216 28 801 737
www.ngi-maghreb.com
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From avi at avimarcus.net  Tue Apr 24 17:40:44 2012
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 24 Apr 2012 16:40:44 +0300
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
Message-ID: <CALNZuwkamd4EBbAkgaFP-foxrYqNjKbTHcVKo2zNsFK0a4fS7w@mail.gmail.com>

Can you paste the actual log and/or sip trace of this? Codec negotiation
often gets complicated, so seeing the log will be very helpful.

-Avi


On Tue, Apr 24, 2012 at 1:19 PM, Cucu Florian <koralu at gmail.com> wrote:

> Hello,
>
> I tried to test the behavior of "bypass media" in FS with the following
> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>
> On each Xlite I set one single codec (that is currently unsupported by FS)
> as it said in wiki page: *"This is useful if you have two end-points that
> need to use a codec that is currently not supported in FreeSWITCH "*
>
> As the FS tutorial said *"Before executing the bridge action you must set
> the "bypass_media" flag to true. "* so I put the following line in my
> dial plan.
>
> <extension name="extension-intercom">
>     <condition field="destination_number" expression="1001">
>        * <action application="set" data="bypass_media=true"/>*
>         <action application="bridge" data="user/1001"/>
>     </condition>
> </extension>
>
>
> <extension name="extension-intercom">
>     <condition field="destination_number" expression="1000">
>         *<action application="set" data="bypass_media=true"/>*
>         <action application="bridge" data="user/1000"/>
>     </condition>
> </extension>
>
> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
> ok. But when I put only codecs that FS do not support(ex.
> iLBC,BroadVoice32) the call failed. What am I missing?
>
> I made this scenario because I need to route via FS some calls with G729r8
> codec and I need no codec negotiation.
>
> Ty
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From krice at freeswitch.org  Tue Apr 24 17:55:35 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 24 Apr 2012 08:55:35 -0500
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
Message-ID: <CBBC1B07.31651%krice@freeswitch.org>

You still need to have the codecs enabled in your FreeSWITCH configuration,
also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
list of other codecs that are just not enabled by default...

K



On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:

> Hello,
> 
> I tried to test the behavior of "bypass media" in FS with the following
> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
> 
> On each Xlite I set one single codec (that is currently unsupported by FS) as
> it said in wiki page: "This is useful if you have two end-points that need to
> use a codec that is currently not supported in FreeSWITCH "
> 
> As the FS tutorial said "Before executing the bridge action you must set the
> "bypass_media" flag to true. " so I put the following line in my dial plan.
> 
> <extension name="extension-intercom">
> ??? <condition field="destination_number" expression="1001">
> ??? ??? <action application="set" data="bypass_media=true"/>
> ??? ??? <action application="bridge" data="user/1001"/>
> ??? </condition>
> </extension>
> 
> 
> <extension name="extension-intercom">
> ??? <condition field="destination_number" expression="1000">
> ??? ??? <action application="set" data="bypass_media=true"/>
> ??? ??? <action application="bridge" data="user/1000"/>
> ??? </condition>
> </extension>
> 
> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was ok.
> But when I put only codecs that FS do not support(ex. iLBC,BroadVoice32) the
> call failed. What am I missing?
> 
> I made this scenario because I need to route via FS some calls with G729r8
> codec and I need no codec negotiation.
> 
> Ty
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From steveu at coppice.org  Tue Apr 24 18:10:45 2012
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 24 Apr 2012 22:10:45 +0800
Subject: [Freeswitch-users] How to enable Hardware DTMF
 detection	(Dahdi/Digium card)
In-Reply-To: <CANmn8agD4c8ZgRTmekjqsqFVMOhj5Ep1yeh15vMb5aMWBDKW7g@mail.gmail.com>
References: <CANmn8agD4c8ZgRTmekjqsqFVMOhj5Ep1yeh15vMb5aMWBDKW7g@mail.gmail.com>
Message-ID: <4F96B465.5010000@coppice.org>

On 04/24/2012 07:46 PM, Haythem Horbit wrote:
> Hi,
>
> We experience some issues to capture SIA ContactID messages from an 
> alarm system, with Freeswitch,
> the problem is that some DTMFs are missed randomly, so that we can't 
> have a correct message
> we use a Digium card 'AEX2400' with DAHDI drivers, the alarm system is 
> connected to an FXO port.
>
> We see in the freeswitch log that software DTMF detection is enabled, 
> may be the issue comes from this fact...
>
> So our question is: how can we enable hardware DTMF detection in DADHI 
> configuration with freeswitch,
> and how to disable software DTMF detection in freeswitch
Does your card have one of those octasic echo canceller modules? Those 
things are a menace.

Steve



From steveu at coppice.org  Tue Apr 24 18:32:03 2012
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 24 Apr 2012 22:32:03 +0800
Subject: [Freeswitch-users] mod_lumenvox ?
In-Reply-To: <CAKV-jFwPUdWD+p0nP9-DriveGvJWmieMSq5VvDFgTacQuaCg4A@mail.gmail.com>
References: <CAGKPtBsd4pcCrNqOXUEiQvxhdn8+__YaRB9tHL-yws9QEYr5ZQ@mail.gmail.com>	<CAKV-jFw0zZUZ75S1h01f92CbzNarA8uGNk7FyU6iUf5shWCA5w@mail.gmail.com>
	<CAKV-jFwPUdWD+p0nP9-DriveGvJWmieMSq5VvDFgTacQuaCg4A@mail.gmail.com>
Message-ID: <4F96B963.8080807@coppice.org>

On 04/24/2012 02:23 PM, Brian Foster wrote:
> A little searching turned up that mod_lumenvox is not and will not be 
> offered. Looks like the company who owns that technology wanted some 
> $$ from the guy who developed the module, so it was taken out of the 
> GIT repo.

Lumenvox seem to have a history of hostility to anyone trying to promote 
or use their products.

Steve



From anthony.minessale at gmail.com  Tue Apr 24 18:41:39 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 24 Apr 2012 09:41:39 -0500
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CBBC1B07.31651%krice@freeswitch.org>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
Message-ID: <CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>

To use bypass from the bypassl_media=true variable you must enable
inbound-late-negotiation in your sofia profile.


On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
> You still need to have the codecs enabled in your FreeSWITCH configuration,
> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
> list of other codecs that are just not enabled by default...
>
> K
>
>
>
>
> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>
> Hello,
>
> I tried to test the behavior of "bypass media" in FS with the following
> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>
> On each Xlite I set one single codec (that is currently unsupported by FS)
> as it said in wiki page: "This is useful if you have two end-points that
> need to use a codec that is currently not supported in FreeSWITCH "
>
> As the FS tutorial said "Before executing the bridge action you must set the
> "bypass_media" flag to true. " so I put the following line in my dial plan.
>
> <extension name="extension-intercom">
> ??? <condition field="destination_number" expression="1001">
> ??? ??? <action application="set" data="bypass_media=true"/>
> ??? ??? <action application="bridge" data="user/1001"/>
> ??? </condition>
> </extension>
>
>
> <extension name="extension-intercom">
> ??? <condition field="destination_number" expression="1000">
> ??? ??? <action application="set" data="bypass_media=true"/>
> ??? ??? <action application="bridge" data="user/1000"/>
> ??? </condition>
> </extension>
>
> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
> ok. But when I put only codecs that FS do not support(ex. iLBC,BroadVoice32)
> the call failed. What am I missing?
>
> I made this scenario because I need to route via FS some calls with G729r8
> codec and I need no codec negotiation.
>
> Ty
>
> ________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From koralu at gmail.com  Tue Apr 24 18:24:56 2012
From: koralu at gmail.com (Cucu Florian)
Date: Tue, 24 Apr 2012 17:24:56 +0300
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CBBC1B07.31651%krice@freeswitch.org>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
Message-ID: <CABvcZya_AqVk2zP=DvccK=BjApg1pu1CkQzsTH5DnfORriuRRA@mail.gmail.com>

Log when G711u not enable:

2012-04-24 17:03:36.107483 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/sip:1001 at 88.88.88.88:56170
[48e4675c-8e16-11e1-ab52-23197b03d6f4]
2012-04-24 17:03:36.567473 [NOTICE] sofia.c:6293 Hangup
sofia/internal/sip:1001 at 88.88.88.88:56170 [CS_CONSUME_MEDIA]
[INCOMPATIBLE_DESTINATION]
2012-04-24 17:03:36.567473 [NOTICE] switch_ivr_originate.c:2459 Cannot
create outgoing channel of type [user] cause:
[INCOMPATIBLE_DESTINATION]
2012-04-24 17:03:36.567473 [INFO] mod_dptools.c:2916 Originate Failed.
 Cause: INCOMPATIBLE_DESTINATION
2012-04-24 17:03:36.567473 [NOTICE] mod_dptools.c:3035 Hangup
sofia/internal/1000 at floflo85.zapto.org [CS_EXECUTE]
[INCOMPATIBLE_DESTINATION]
2012-04-24 17:03:36.567473 [NOTICE] switch_core_session.c:1398 Session
2 (sofia/internal/sip:1001 at 88.88.88.88:56170) Ended
2012-04-24 17:03:36.567473 [NOTICE] switch_core_session.c:1400 Close
Channel sofia/internal/sip:1001 at 88.88.88.88:56170 [CS_DESTROY]
2012-04-24 17:03:36.588379 [NOTICE] switch_core_session.c:1398 Session
1 (sofia/internal/1000 at floflo85.zapto.org) Ended
2012-04-24 17:03:36.588379 [NOTICE] switch_core_session.c:1400 Close
Channel sofia/internal/1000 at floflo85.zapto.org [CS_DESTROY]
2012-04-24 17:04:49.567473 [NOTICE] switch_channel.c:926 New Channel
sofia/internal/1000 at floflo85.zapto.org
[74ad1b7c-8e16-11e1-ab57-23197b03d6f4]

@Ken Rice: Where should I enable the codecs?What if the codec is not
supported by FS? What I want to do is that codecs negotiation be made
by end point wihout any action by FS.

Ty

On 4/24/12, Ken Rice <krice at freeswitch.org> wrote:
> You still need to have the codecs enabled in your FreeSWITCH configuration,
> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
> list of other codecs that are just not enabled by default...
>
> K
>
>
>
> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>
>> Hello,
>>
>> I tried to test the behavior of "bypass media" in FS with the following
>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>>
>> On each Xlite I set one single codec (that is currently unsupported by FS)
>> as
>> it said in wiki page: "This is useful if you have two end-points that need
>> to
>> use a codec that is currently not supported in FreeSWITCH "
>>
>> As the FS tutorial said "Before executing the bridge action you must set
>> the
>> "bypass_media" flag to true. " so I put the following line in my dial
>> plan.
>>
>> <extension name="extension-intercom">
>> ??? <condition field="destination_number" expression="1001">
>> ??? ??? <action application="set" data="bypass_media=true"/>
>> ??? ??? <action application="bridge" data="user/1001"/>
>> ??? </condition>
>> </extension>
>>
>>
>> <extension name="extension-intercom">
>> ??? <condition field="destination_number" expression="1000">
>> ??? ??? <action application="set" data="bypass_media=true"/>
>> ??? ??? <action application="bridge" data="user/1000"/>
>> ??? </condition>
>> </extension>
>>
>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
>> ok.
>> But when I put only codecs that FS do not support(ex. iLBC,BroadVoice32)
>> the
>> call failed. What am I missing?
>>
>> I made this scenario because I need to route via FS some calls with G729r8
>> codec and I need no codec negotiation.
>>
>> Ty
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>


From koralu at gmail.com  Tue Apr 24 19:55:43 2012
From: koralu at gmail.com (Cucu Florian)
Date: Tue, 24 Apr 2012 18:55:43 +0300
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
	<CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>
Message-ID: <CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>

Thank you for the replies but didn't work.

I uncomment late-negotioation parameter in sip_profile/internal.xml
<param name="inbound-late-negotiation" value="true"/>

I put <action application="set" data="bypass_media=true"/> in my dial plan.

I set for both Xlite only iLBC codec. The logs are:

2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf
send/recv payload to 101
2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848
(sofia/internal/1001 at floflo85.zapto.org) Callstate Change DOWN ->
HANGUP
2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/1001 at floflo85.zapto.org) Running State Change
CS_HANGUP
2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup
sofia/internal/1001 at floflo85.zapto.org [CS_NEW]
[INCOMPATIBLE_DESTINATION]

 When I set for just one Xlite iLBC and also G711 the call is ok.

Ty.

On 4/24/12, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> To use bypass from the bypassl_media=true variable you must enable
> inbound-late-negotiation in your sofia profile.
>
>
> On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>> You still need to have the codecs enabled in your FreeSWITCH
>> configuration,
>> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
>> list of other codecs that are just not enabled by default...
>>
>> K
>>
>>
>>
>>
>> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>>
>> Hello,
>>
>> I tried to test the behavior of "bypass media" in FS with the following
>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>>
>> On each Xlite I set one single codec (that is currently unsupported by FS)
>> as it said in wiki page: "This is useful if you have two end-points that
>> need to use a codec that is currently not supported in FreeSWITCH "
>>
>> As the FS tutorial said "Before executing the bridge action you must set
>> the
>> "bypass_media" flag to true. " so I put the following line in my dial
>> plan.
>>
>> <extension name="extension-intercom">
>> ??? <condition field="destination_number" expression="1001">
>> ??? ??? <action application="set" data="bypass_media=true"/>
>> ??? ??? <action application="bridge" data="user/1001"/>
>> ??? </condition>
>> </extension>
>>
>>
>> <extension name="extension-intercom">
>> ??? <condition field="destination_number" expression="1000">
>> ??? ??? <action application="set" data="bypass_media=true"/>
>> ??? ??? <action application="bridge" data="user/1000"/>
>> ??? </condition>
>> </extension>
>>
>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
>> ok. But when I put only codecs that FS do not support(ex.
>> iLBC,BroadVoice32)
>> the call failed. What am I missing?
>>
>> I made this scenario because I need to route via FS some calls with G729r8
>> codec and I need no codec negotiation.
>>
>> Ty
>>
>> ________________________________
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Tue Apr 24 20:16:57 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 24 Apr 2012 11:16:57 -0500
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
	<CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>
	<CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>
Message-ID: <CAKbxfG8ApUNZBoDMyeeFA-5VdgYGbd_AGgdXSAFOx1dMz00SSw@mail.gmail.com>

did you set it first in your dialplan before anything else and make
sure your dialplan does nothing at all that reqires media.
Ideally nothing but setting the var and the bridge line.

<action application="set" data="bypass_media=true"/>
<action application="bridge" data="user/1004"/>

if you do anything at all that requires media from FS it will about
the bypass and establish media from FS.

It works, once you tell yourself that and look for how you have
misconfigured it, then you will have success.



On Tue, Apr 24, 2012 at 10:55 AM, Cucu Florian <koralu at gmail.com> wrote:
> Thank you for the replies but didn't work.
>
> I uncomment late-negotioation parameter in sip_profile/internal.xml
> <param name="inbound-late-negotiation" value="true"/>
>
> I put <action application="set" data="bypass_media=true"/> in my dial plan.
>
> I set for both Xlite only iLBC codec. The logs are:
>
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf
> send/recv payload to 101
> 2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848
> (sofia/internal/1001 at floflo85.zapto.org) Callstate Change DOWN ->
> HANGUP
> 2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/1001 at floflo85.zapto.org) Running State Change
> CS_HANGUP
> 2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup
> sofia/internal/1001 at floflo85.zapto.org [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
>
> ?When I set for just one Xlite iLBC and also G711 the call is ok.
>
> Ty.
>
> On 4/24/12, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> To use bypass from the bypassl_media=true variable you must enable
>> inbound-late-negotiation in your sofia profile.
>>
>>
>> On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> You still need to have the codecs enabled in your FreeSWITCH
>>> configuration,
>>> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
>>> list of other codecs that are just not enabled by default...
>>>
>>> K
>>>
>>>
>>>
>>>
>>> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>> I tried to test the behavior of "bypass media" in FS with the following
>>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>>>
>>> On each Xlite I set one single codec (that is currently unsupported by FS)
>>> as it said in wiki page: "This is useful if you have two end-points that
>>> need to use a codec that is currently not supported in FreeSWITCH "
>>>
>>> As the FS tutorial said "Before executing the bridge action you must set
>>> the
>>> "bypass_media" flag to true. " so I put the following line in my dial
>>> plan.
>>>
>>> <extension name="extension-intercom">
>>> ??? <condition field="destination_number" expression="1001">
>>> ??? ??? <action application="set" data="bypass_media=true"/>
>>> ??? ??? <action application="bridge" data="user/1001"/>
>>> ??? </condition>
>>> </extension>
>>>
>>>
>>> <extension name="extension-intercom">
>>> ??? <condition field="destination_number" expression="1000">
>>> ??? ??? <action application="set" data="bypass_media=true"/>
>>> ??? ??? <action application="bridge" data="user/1000"/>
>>> ??? </condition>
>>> </extension>
>>>
>>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
>>> ok. But when I put only codecs that FS do not support(ex.
>>> iLBC,BroadVoice32)
>>> the call failed. What am I missing?
>>>
>>> I made this scenario because I need to route via FS some calls with G729r8
>>> codec and I need no codec negotiation.
>>>
>>> Ty
>>>
>>> ________________________________
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From haythem.horbit at gmail.com  Tue Apr 24 20:08:47 2012
From: haythem.horbit at gmail.com (Haythem Horbit)
Date: Tue, 24 Apr 2012 18:08:47 +0200
Subject: [Freeswitch-users] How to enable Hardware DTMF detection
 (Dahdi/Digium card)
In-Reply-To: <4F96B465.5010000@coppice.org>
References: <CANmn8agD4c8ZgRTmekjqsqFVMOhj5Ep1yeh15vMb5aMWBDKW7g@mail.gmail.com>
	<4F96B465.5010000@coppice.org>
Message-ID: <CANmn8ahkTsz-VoqQP3H067nWQHD-GS7nVvs0E2YQB4TYNs-ejg@mail.gmail.com>

I had disabled echo canceller modules but the problem didn't resolved
 ( wctdm24xxp 0000:25:08.0: VPM: Support Disabled   see :
http://kb.digium.com/articles/FAQ/How-do-I-disable-echo-cancellation-module-without-removing-the-hardware?retURL=%2F&popup=false
)
I'am using mod_freetdm   (dahdi)

2012/4/24 Steve Underwood <steveu at coppice.org>

> On 04/24/2012 07:46 PM, Haythem Horbit wrote:
> > Hi,
> >
> > We experience some issues to capture SIA ContactID messages from an
> > alarm system, with Freeswitch,
> > the problem is that some DTMFs are missed randomly, so that we can't
> > have a correct message
> > we use a Digium card 'AEX2400' with DAHDI drivers, the alarm system is
> > connected to an FXO port.
> >
> > We see in the freeswitch log that software DTMF detection is enabled,
> > may be the issue comes from this fact...
> >
> > So our question is: how can we enable hardware DTMF detection in DADHI
> > configuration with freeswitch,
> > and how to disable software DTMF detection in freeswitch
> Does your card have one of those octasic echo canceller modules? Those
> things are a menace.
>
> Steve
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Haythem HORBIT
Responsable d?veloppement
NGI Maghreb*
53, rue 8601 - 2035 Charguia I
Tunis - TUNISIE

T?l.    : +216 71 808 565
Fax    : +216 71 808 395
Gsm   : +216 28 801 737
www.ngi-maghreb.com




P  Before printing, think about ENVIRONMENTAL responsibility
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From peter.olsson at visionutveckling.se  Tue Apr 24 20:32:56 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 24 Apr 2012 16:32:56 +0000
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
	<CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>,
	<CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>
Message-ID: <75180303-8636-49D8-B65C-801FBB58E261@visionutveckling.se>

Did you restart the sofia profile after the change?

/Peter

24 apr 2012 kl. 18:03 skrev "Cucu Florian" <koralu at gmail.com>:

> Thank you for the replies but didn't work.
> 
> I uncomment late-negotioation parameter in sip_profile/internal.xml
> <param name="inbound-late-negotiation" value="true"/>
> 
> I put <action application="set" data="bypass_media=true"/> in my dial plan.
> 
> I set for both Xlite only iLBC codec. The logs are:
> 
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf
> send/recv payload to 101
> 2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848
> (sofia/internal/1001 at floflo85.zapto.org) Callstate Change DOWN ->
> HANGUP
> 2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/1001 at floflo85.zapto.org) Running State Change
> CS_HANGUP
> 2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup
> sofia/internal/1001 at floflo85.zapto.org [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
> 
> When I set for just one Xlite iLBC and also G711 the call is ok.
> 
> Ty.
> 
> On 4/24/12, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> To use bypass from the bypassl_media=true variable you must enable
>> inbound-late-negotiation in your sofia profile.
>> 
>> 
>> On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> You still need to have the codecs enabled in your FreeSWITCH
>>> configuration,
>>> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
>>> list of other codecs that are just not enabled by default...
>>> 
>>> K
>>> 
>>> 
>>> 
>>> 
>>> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>>> 
>>> Hello,
>>> 
>>> I tried to test the behavior of "bypass media" in FS with the following
>>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>>> 
>>> On each Xlite I set one single codec (that is currently unsupported by FS)
>>> as it said in wiki page: "This is useful if you have two end-points that
>>> need to use a codec that is currently not supported in FreeSWITCH "
>>> 
>>> As the FS tutorial said "Before executing the bridge action you must set
>>> the
>>> "bypass_media" flag to true. " so I put the following line in my dial
>>> plan.
>>> 
>>> <extension name="extension-intercom">
>>>     <condition field="destination_number" expression="1001">
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge" data="user/1001"/>
>>>     </condition>
>>> </extension>
>>> 
>>> 
>>> <extension name="extension-intercom">
>>>     <condition field="destination_number" expression="1000">
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge" data="user/1000"/>
>>>     </condition>
>>> </extension>
>>> 
>>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
>>> ok. But when I put only codecs that FS do not support(ex.
>>> iLBC,BroadVoice32)
>>> the call failed. What am I missing?
>>> 
>>> I made this scenario because I need to route via FS some calls with G729r8
>>> codec and I need no codec negotiation.
>>> 
>>> Ty
>>> 
>>> ________________________________
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4f96cbdf32769280115710!
> 


From amilkhanzada at gmail.com  Tue Apr 24 20:34:07 2012
From: amilkhanzada at gmail.com (Amil)
Date: Tue, 24 Apr 2012 09:34:07 -0700
Subject: [Freeswitch-users] Using Softphone on same machine as
	FreeSwitch?
In-Reply-To: <loom.20120410T213858-112@post.gmane.org>
References: <33649239.post@talk.nabble.com>
	<CAHKqMb635kBHY4-y-bh=4kYSL7xYmAHq8jqiJBkktwMCAMED7Q@mail.gmail.com>
	<loom.20120410T213858-112@post.gmane.org>
Message-ID: <CAOTFBbe1cbfjNhZ5FZj41VfdagRAkhX6X7TfD1N9zp+m5_UE6g@mail.gmail.com>

Hello all,

I finally got it working. It seems that Twinkle is pretty buggy in terms of
SIP (at least when trying to do everything on one machine) and this seems
to be why it was giving me such erratic behavior.
So, I used pidgin's SIMPLE instead and didn't run into any issues.

Also, my default.xml had some problems like having malformed multiple
conditions in each extension.

Thanks again,
Amil
On Apr 10, 2012 1:00 PM, "LearningFS" <amilkhanzada at gmail.com> wrote:

> Nandy Dagondon <gcd <at> i.ph> writes:
>
> >
> > the x-lite is okay but IMHO add at least 2 real phones (IP or ATA) for
> complete testing e.g. conference calls, call forwarding, etc.
> >
>
> Hi Nandy,
>
> Thanks for the response. I finally got a softphone working by:
> 1. In "conf/vars.xml", changing 'data="$${local_ip_v4}"' to
> 'data="domain=127.0.0.1"'
> 2. Downloading Twinkle via Ubuntu package manager
> 3. Changing Twinkle to use Port 5062 in System Settings->Network->Sip Port
> (this
> was necessary because FS was locking port 5060 as everything is on the same
> machine)
> 4. Setting up my SIP account in Twinkle using my IP address shown as "inet
> addr:10.0.2.15" after running ifconfig
> 5. Adding a contact with phone number "sip:1019 at 10.0.2.15"
> 6. Right clicking this contact and sending an instant message
>
>
> However, I am getting strange problems now. Twinkle can make phone calls,
> but it
> crashes when I try to end a phone call.
> Twinkle can send text messages and freeswitch shows "[INFO] mod_sms.c:299
> Processing text message 1004->1019 in context default" but my default.xml
> script
> is not executed (as seen on http://wiki.freeswitch.org/wiki/Mod_sms). I
> don't
> get any response saying "Hello, you said hello"
>
>
> Any ideas?
> I'll try digging in to this more, but I really think this has something to
> do
> with me trying to run everything on the same machine. Is there any setting
> in
> FreeSwitch to allow this?
>
> Thanks,
> Amil
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From steveu at coppice.org  Tue Apr 24 21:02:21 2012
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 25 Apr 2012 01:02:21 +0800
Subject: [Freeswitch-users] How to enable Hardware DTMF detection
 (Dahdi/Digium card)
In-Reply-To: <CANmn8ahkTsz-VoqQP3H067nWQHD-GS7nVvs0E2YQB4TYNs-ejg@mail.gmail.com>
References: <CANmn8agD4c8ZgRTmekjqsqFVMOhj5Ep1yeh15vMb5aMWBDKW7g@mail.gmail.com>	<4F96B465.5010000@coppice.org>
	<CANmn8ahkTsz-VoqQP3H067nWQHD-GS7nVvs0E2YQB4TYNs-ejg@mail.gmail.com>
Message-ID: <4F96DC9D.2060504@coppice.org>

On 04/25/2012 12:08 AM, Haythem Horbit wrote:
> I had disabled echo canceller modules but the problem didn't resolved
>  ( wctdm24xxp 0000:25:08.0: VPM: Support Disabled   see : 
> http://kb.digium.com/articles/FAQ/How-do-I-disable-echo-cancellation-module-without-removing-the-hardware?retURL=%2F&popup=false 
> <http://kb.digium.com/articles/FAQ/How-do-I-disable-echo-cancellation-module-without-removing-the-hardware?retURL=%2F&popup=false>)
> I'am using mod_freetdm   (dahdi)
Try relaxing the DTMF twist thresholds. The need for that seems to 
typically be a problem with calls from cellular systems, but it can 
happen elsewhere when something is screwing badly with the frequency 
response. Someone put changes into git recently so you can change the 
setting through channel variables.

Steve
>
> 2012/4/24 Steve Underwood <steveu at coppice.org <mailto:steveu at coppice.org>>
>
>     On 04/24/2012 07:46 PM, Haythem Horbit wrote:
>     > Hi,
>     >
>     > We experience some issues to capture SIA ContactID messages from an
>     > alarm system, with Freeswitch,
>     > the problem is that some DTMFs are missed randomly, so that we can't
>     > have a correct message
>     > we use a Digium card 'AEX2400' with DAHDI drivers, the alarm
>     system is
>     > connected to an FXO port.
>     >
>     > We see in the freeswitch log that software DTMF detection is
>     enabled,
>     > may be the issue comes from this fact...
>     >
>     > So our question is: how can we enable hardware DTMF detection in
>     DADHI
>     > configuration with freeswitch,
>     > and how to disable software DTMF detection in freeswitch
>     Does your card have one of those octasic echo canceller modules? Those
>     things are a menace.
>
>     Steve
>



From bedgar at vseinc.com  Tue Apr 24 21:56:20 2012
From: bedgar at vseinc.com (bedgar at vseinc.com)
Date: Tue, 24 Apr 2012 13:56:20 -0400
Subject: [Freeswitch-users] FreeSWITCH and Sangoma - Robbed Bit Channel
 Associated Signaling (CAS)
Message-ID: <333789DE5C38474EB3A478A538F4EBAB0A333F0025@prod-exch01.corp.vseinc.com>

Are there any tricks to getting Robbed Bit Channel Associated Signaling (CAS) working properly with FS (legacy 1.06 nor latest git seem to show desired results) and Sangoma boards.  I am using model A102, REV 1.1.  DTMF does not seem to be recognized in the call stream so IVR does not respond.  SIP calls work fine.

Brian C. Edgar, Jr.

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From koralu at gmail.com  Wed Apr 25 01:15:17 2012
From: koralu at gmail.com (Cucu Florian)
Date: Wed, 25 Apr 2012 00:15:17 +0300
Subject: [Freeswitch-users] Bypass Media not work
In-Reply-To: <75180303-8636-49D8-B65C-801FBB58E261@visionutveckling.se>
References: <CABvcZybpVG6Qxk2MkhxQ7re8Asbfq_krLFs+4MgWsi6X85QoYQ@mail.gmail.com>
	<CBBC1B07.31651%krice@freeswitch.org>
	<CAKbxfG-VtHxis5USySQm2XZU-c7nft2encr+SSAdtZ3pSVdhrw@mail.gmail.com>
	<CABvcZyaKup03Yw3mvweannWiPww+QmTzRr+Jc86i=60R6Kdnag@mail.gmail.com>
	<75180303-8636-49D8-B65C-801FBB58E261@visionutveckling.se>
Message-ID: <CABvcZyax2Xm-EW1NQzt3MUgGbo3u4Qc_whgkrXqQV4Rsj26iJA@mail.gmail.com>

It works. Thank you Anthony for the advice with late-negotiation and thank
you Peter for your replay. Me stupid, I forgot to restart sofia profile.

On Tue, Apr 24, 2012 at 7:32 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Did you restart the sofia profile after the change?
>
> /Peter
>
> 24 apr 2012 kl. 18:03 skrev "Cucu Florian" <koralu at gmail.com>:
>
> > Thank you for the replies but didn't work.
> >
> > I uncomment late-negotioation parameter in sip_profile/internal.xml
> > <param name="inbound-late-negotiation" value="true"/>
> >
> > I put <action application="set" data="bypass_media=true"/> in my dial
> plan.
> >
> > I set for both Xlite only iLBC codec. The logs are:
> >
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> > Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
> > 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf
> > send/recv payload to 101
> > 2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848
> > (sofia/internal/1001 at floflo85.zapto.org) Callstate Change DOWN ->
> > HANGUP
> > 2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362
> > (sofia/internal/1001 at floflo85.zapto.org) Running State Change
> > CS_HANGUP
> > 2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup
> > sofia/internal/1001 at floflo85.zapto.org [CS_NEW]
> > [INCOMPATIBLE_DESTINATION]
> >
> > When I set for just one Xlite iLBC and also G711 the call is ok.
> >
> > Ty.
> >
> > On 4/24/12, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >> To use bypass from the bypassl_media=true variable you must enable
> >> inbound-late-negotiation in your sofia profile.
> >>
> >>
> >> On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
> >>> You still need to have the codecs enabled in your FreeSWITCH
> >>> configuration,
> >>> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along
> with a
> >>> list of other codecs that are just not enabled by default...
> >>>
> >>> K
> >>>
> >>>
> >>>
> >>>
> >>> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
> >>>
> >>> Hello,
> >>>
> >>> I tried to test the behavior of "bypass media" in FS with the following
> >>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
> >>>
> >>> On each Xlite I set one single codec (that is currently unsupported by
> FS)
> >>> as it said in wiki page: "This is useful if you have two end-points
> that
> >>> need to use a codec that is currently not supported in FreeSWITCH "
> >>>
> >>> As the FS tutorial said "Before executing the bridge action you must
> set
> >>> the
> >>> "bypass_media" flag to true. " so I put the following line in my dial
> >>> plan.
> >>>
> >>> <extension name="extension-intercom">
> >>>     <condition field="destination_number" expression="1001">
> >>>         <action application="set" data="bypass_media=true"/>
> >>>         <action application="bridge" data="user/1001"/>
> >>>     </condition>
> >>> </extension>
> >>>
> >>>
> >>> <extension name="extension-intercom">
> >>>     <condition field="destination_number" expression="1000">
> >>>         <action application="set" data="bypass_media=true"/>
> >>>         <action application="bridge" data="user/1000"/>
> >>>     </condition>
> >>> </extension>
> >>>
> >>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls
> was
> >>> ok. But when I put only codecs that FS do not support(ex.
> >>> iLBC,BroadVoice32)
> >>> the call failed. What am I missing?
> >>>
> >>> I made this scenario because I need to route via FS some calls with
> G729r8
> >>> codec and I need no codec negotiation.
> >>>
> >>> Ty
> >>>
> >>> ________________________________
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > !DSPAM:4f96cbdf32769280115710!
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From rml at tollfreeforwarding.com  Wed Apr 25 02:39:23 2012
From: rml at tollfreeforwarding.com (RaviRaj Mulasa)
Date: Tue, 24 Apr 2012 22:39:23 +0000
Subject: [Freeswitch-users] HTTPS and mod_http_cache
Message-ID: <8B94625BC339264DBA61E314BE9EC2CF2E9CAFB5@EXCH125.IFN.com>

Hi

I am trying to prefetch a file to play using mod_http_cache.
The URL @ which the file is located uses HTTPS instead of HTTP

HTTPS
freeswitch at myplayground> http_prefetch https://myfiles/file_1.wav
USAGE: <url>


HTTP
freeswitch at myplayground> http_prefetch http://myfiles/file_1.wav
+OK

2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:396 Locked cache
2012-04-24 08:25:48.108625 [INFO] mod_http_cache.c:476 Cache MISS: size = 0 (0 MB), hit ratio = 0/1
2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:541 Adding http://myfiles/file_1.wav(/tmp/http_cache/c4/9828e2-8e21-11e1-9e33-017c98d37fa6.wav) to cache index 0
2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:407 Unlocked cache
2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:728 opening /tmp/http_cache/



Please let me know how can I
HTTP_PREFETCH
HTTP_GET                           using a URL that uses HTTPS.

Thanks
RaviRaj.


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From cmrienzo at gmail.com  Wed Apr 25 04:25:10 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 24 Apr 2012 20:25:10 -0400
Subject: [Freeswitch-users] HTTPS and mod_http_cache
In-Reply-To: <8B94625BC339264DBA61E314BE9EC2CF2E9CAFB5@EXCH125.IFN.com>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CAFB5@EXCH125.IFN.com>
Message-ID: <CAD-qRkjncA3w7VhatWdW4EBx-2i8cJe-17ZiAQvp_NS7MzCZDA@mail.gmail.com>

Sync with latest version of FS; I added https support.  There is a
cacert.pem file in mod_http_cache/conf directory that you'll want to put in
freeswitch/conf directory.

The following new params are available in http_cache.conf

ssl-cacert : this is the absolute path to the cacert.pem file.  default is
/usr/local/freeswitch/conf/cacert.pem
ssl-verifypeer : if false the cert is not verified
ssl-verifyhost : if false the host name does not have to match the cert

Let me know if there are any problems.

Chris


On Tue, Apr 24, 2012 at 6:39 PM, RaviRaj Mulasa
<rml at tollfreeforwarding.com>wrote:

> Hi****
>
> ** **
>
> I am trying to prefetch a file to play using mod_http_cache.****
>
> The URL @ which the file is located uses *HTTPS instead of HTTP*****
>
> ** **
>
> *HTTPS*
>
> freeswitch at myplayground> http_prefetch https://myfiles/file_1.wav****
>
> USAGE: <url>****
>
> ** **
>
> ** **
>
> *HTTP*
>
> freeswitch at myplayground> http_prefetch http://myfiles/file_1.wav****
>
> +OK****
>
> ** **
>
> 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:396 Locked cache****
>
> 2012-04-24 08:25:48.108625 [INFO] mod_http_cache.c:476 Cache MISS: size =
> 0 (0 MB), hit ratio = 0/1****
>
> 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:541 Adding
> http://myfiles/file_1.wav(/tmp/http_cache/c4/9828e2-8e21-11e1-9e33-017c98d37fa6.wav)to cache index 0
> ****
>
> 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:407 Unlocked cache****
>
> 2012-04-24 08:25:48.108625 [DEBUG] mod_http_cache.c:728 opening
> /tmp/http_cache/****
>
> ** **
>
> ** **
>
> ** **
>
> Please let me know how can I ****
>
> HTTP_PREFETCH ****
>
> HTTP_GET                           using a URL that uses HTTPS.****
>
> ** **
>
> Thanks****
>
> RaviRaj.****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anita.hall at simmortel.com  Wed Apr 25 12:10:02 2012
From: anita.hall at simmortel.com (Anita Hall)
Date: Wed, 25 Apr 2012 13:40:02 +0530
Subject: [Freeswitch-users] play_and_detect_speech errors
Message-ID: <CAGKPtBs=NABydjue-gxLNj6-G79qU5Q=+AZFXJ3m-tY=Y9UATw@mail.gmail.com>

Hi

I was using play_and_detect_speech with Nuance Speech Server 5 and it
generally works great! Much thanks to Chris Rienzo for this cool App :)

There are 2 cases where I need help to understand what is happening.

1. In general my SIP connection between mod_unimrcp and Nuance works fine
with all the calls answered. But in 3 repeated cases, the SIP connection
was not established as a result of which the call got hungup.

Is there any graceful way in play_and_detect_speech (like some channel
variable set) which can be used to find if the SIP call has been answered
by Nuance or not.

Like in this case, I got

2012-04-20 13:20:47.955516 [DEBUG] apt_consumer_task.c:97 () Wait for
Messages [MRCP Client]
  7430 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
() Receive SIP Event [nua_i_state] Status 0 INVITE sent
  7431 2012-04-20 13:20:47.955516 [NOTICE] mrcp_sofiasip_client_agent.c:522
() SIP Call State 0x7f4f04188270 [calling]
  7432 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
() Receive SIP Event [nua_r_invite] Status 503 Service Unavailable
  7433 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
() Receive SIP Event [nua_i_state] Status 503 Service Unavailable
  7434 2012-04-20 13:20:47.955516 [NOTICE] mrcp_sofiasip_client_agent.c:522
() SIP Call State 0x7f4f04188270 [terminated]
  7435 2012-04-20 13:20:47.955516 [DEBUG] apt_task.c:278 () Process Message
[MRCP Client] [1;4]
  7436 2012-04-20 13:20:47.955516 [INFO] mrcp_client_session.c:535 (ASR-0)
Raise App Response 0x7f4f04188270 <new> [2] FAILURE [2]
  7437 2012-04-20 13:20:47.955516 [ERR] mod_unimrcp.c:1860 (ASR-0)
RECOGNIZER channel error!
  7438 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1866 Terminating
MRCP session
  7439 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1541 (ASR-0) CLOSED
==> ERROR






2. After performing a recognition with a low confidence score, the call got
disconnected but FreeSWITCH could not detect the call disconnection and
repeatedly gave the following debug message in the log till it flooded
hundreds of log files.

switch_core_session.c:2102 FreeTDM/1:2/4948608 Channel is hungup and
application play_and_detect_speech does not have the zombie_exec flag.

During this "show calls" on fs_cli will crash fs_cli

I had to shutdown freeswitch.






regards,
Anita
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From Adam.Lappe at qsc.de  Wed Apr 25 13:21:57 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 25 Apr 2012 11:21:57 +0200
Subject: [Freeswitch-users] park bridged call legs
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E450@CGNMSG01.oc.qsc.de>

Hi all,

is there a way to park two call legs after I bridged them?
When I use:
uuid_bridge <uuid_1> <uuid_2>
and then
uuid_park <uuid_1>
Freeswitch kills <uuid_2> with NORMAL CLEARING.

I tried to use park_after_bridge but this is not useful imo
since it parks every normal call when any1 hangs up.
Any ideas?

Thanks,
Adam
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From peter.olsson at visionutveckling.se  Wed Apr 25 13:43:22 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 25 Apr 2012 09:43:22 +0000
Subject: [Freeswitch-users] park bridged call legs
Message-ID: <1FFF97C269757C458224B7C895F35F1509904E@cantor.std.visionutv.se>

uuid_transfer should work for this. It can handle the flag "-both", which can transfer both legs somewhere.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Lappe, Adam
Skickat: den 25 april 2012 11:22
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] park bridged call legs

Hi all,

is there a way to park two call legs after I bridged them?
When I use:
uuid_bridge <uuid_1> <uuid_2>
and then
uuid_park <uuid_1>
Freeswitch kills <uuid_2> with NORMAL CLEARING.

I tried to use park_after_bridge but this is not useful imo
since it parks every normal call when any1 hangs up.
Any ideas?

Thanks,
Adam
!DSPAM:4f97c0f632765009093151!
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From joohny at mail.ru  Wed Apr 25 16:34:55 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Wed, 25 Apr 2012 16:34:55 +0400
Subject: [Freeswitch-users] =?utf-8?q?autoanswer_by_re-INVITE?=
Message-ID: <E1SN1Qt-00009z-ST.joohny-mail-ru@f250.mail.ru>

Hi!
I can send a new call with autoanswer by
<action application="export" data="sip_auto_answer=true"/>
It works well.
But I need some different feature
I want to "re-INVITE" existing call with autoanswer option, how can I do this?
Or may be somebody knows, how can I make Cisco SPA508G to answer the call with any command from FreeSWITCH. It will be very usefull with headset to answer calls by pressing some buttons with keybord or mouse on PC.
From Russia with Love :)
Evginey? ????????? ???????.

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From joohny at mail.ru  Wed Apr 25 17:02:13 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Wed, 25 Apr 2012 17:02:13 +0400
Subject: [Freeswitch-users] =?utf-8?q?mod=5Fcallcenter_and_prompts_to_memb?=
	=?utf-8?q?ers?=
Message-ID: <E1SN1rJ-00069x-V7.joohny-mail-ru@f215.mail.ru>

Hello.
I try to make prompting script for queues. 
So I want to know who will be first, second ... last in
freeswitch at 8.8.8.8@default> callcenter_config queue list members 3001
queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state
3001|single_box|cd025909-0751-4f46-9208-34302eb2d34d|392b4df4-45fa-4b1b-8e42-d15117edce5e|0000000000||1335357680|1335357685|0|0|0|5|0|||Waiting
3001|single_box|88006efb-f80b-47fd-a4d6-1c711483d3d3|92e77fde-f2ce-45d7-885b-7475b3ef9b48|0000000000||1335357683|1335357687|0|0|0|4|0|||Waiting
3001|single_box|c6c74d6a-092c-49a4-9242-abba925c1523|69db31f3-786d-4f2a-ae2b-3f5954c0e4c4|0000000000||1335357684|1335357689|0|0|0|5|0|||Waiting
3001|single_box|a2509a89-5717-44cc-8382-c5718f1c9813|8f692a80-6680-4eff-8692-9f71f1843508|0000000000||1335357685|1335357690|0|0|0|5|0|||Waiting
3001|single_box|38dd8346-97cf-4d3a-9e8b-773103e4657f|382e0341-cc24-48de-84c2-d27583bd8115|0000000000||1335357687|1335357692|0|0|0|5|0|||Waiting
3001|single_box|beba7fc8-2eb8-4ff6-bcb3-c08599e0dc19|56e2c2f9-c458-4df6-b3eb-dae4b9ae0a5a|0000000000||1335357688|1335357693|0|0|0|5|0|||Waiting
3001|single_box|c4552675-2be1-4e12-8ca5-a9d20a421be2|8d47431f-c7da-4f4a-9074-ebbbee6a0710|0000000000||1335357690|1335357694|0|0|0|4|0|||Waiting
3001|single_box|89b0ada0-44fb-4f65-98ec-182589fe3fe9|f7263ffd-05ff-4184-a6fe-2aaede34692f|0000000000||1335357691|1335357696|0|0|0|5|0|||Waiting
3001|single_box|72b2bd2e-87fb-4912-a190-fda98ee8cef8|9cede828-db7c-48fb-85f4-6cbd2069c264|0000000000||1335357692|1335357697|0|0|0|5|0|||Waiting
+OK

How can I gues It? Or there is another way to solve this issue?
I want to prompt something like: you are third in a queue.


From Russia with Love :)
Evginey? 
? ????????? ???????.

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From Adam.Lappe at qsc.de  Wed Apr 25 17:03:18 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 25 Apr 2012 15:03:18 +0200
Subject: [Freeswitch-users] park bridged call legs
In-Reply-To: <1FFF97C269757C458224B7C895F35F1509904E@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1509904E@cantor.std.visionutv.se>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E453@CGNMSG01.oc.qsc.de>

I guess I use the wrong syntax.
uuid_transfer <uuid> -both park
isn?t working. Any advice?

Thanks,
Adam

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson
Gesendet: Mittwoch, 25. April 2012 11:43
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] park bridged call legs

uuid_transfer should work for this. It can handle the flag ?-both?, which can transfer both legs somewhere.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Lappe, Adam
Skickat: den 25 april 2012 11:22
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] park bridged call legs

Hi all,

is there a way to park two call legs after I bridged them?
When I use:
uuid_bridge <uuid_1> <uuid_2>
and then
uuid_park <uuid_1>
Freeswitch kills <uuid_2> with NORMAL CLEARING.

I tried to use park_after_bridge but this is not useful imo
since it parks every normal call when any1 hangs up.
Any ideas?

Thanks,
Adam
!DSPAM:4f97c0f632765009093151!
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From peter.olsson at visionutveckling.se  Wed Apr 25 17:23:01 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 25 Apr 2012 13:23:01 +0000
Subject: [Freeswitch-users] park bridged call legs
Message-ID: <1FFF97C269757C458224B7C895F35F15099375@cantor.std.visionutv.se>

Try this, I'm not sure if it works though.. :)

uuid_transfer <uuid> -both park inline

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Lappe, Adam
Skickat: den 25 april 2012 15:03
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] park bridged call legs

I guess I use the wrong syntax.
uuid_transfer <uuid> -both park
isn't working. Any advice?

Thanks,
Adam

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> Im Auftrag von Peter Olsson
Gesendet: Mittwoch, 25. April 2012 11:43
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] park bridged call legs

uuid_transfer should work for this. It can handle the flag "-both", which can transfer both legs somewhere.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> F?r Lappe, Adam
Skickat: den 25 april 2012 11:22
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] park bridged call legs

Hi all,

is there a way to park two call legs after I bridged them?
When I use:
uuid_bridge <uuid_1> <uuid_2>
and then
uuid_park <uuid_1>
Freeswitch kills <uuid_2> with NORMAL CLEARING.

I tried to use park_after_bridge but this is not useful imo
since it parks every normal call when any1 hangs up.
Any ideas?

Thanks,
Adam

!DSPAM:4f97f4cc32761721410779!
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From andrew.paul85 at gmail.com  Wed Apr 25 16:14:04 2012
From: andrew.paul85 at gmail.com (Andrew Paul)
Date: Wed, 25 Apr 2012 17:44:04 +0530
Subject: [Freeswitch-users] One way Audio
Message-ID: <CAMX4_Qkz3e1AU4pyUgdgz8nnorEG6wo-1yjakTk+cAsZdhw+9A@mail.gmail.com>

Hai,
I have a freeswitch setup  in that one asterisk pbx registered. For this
setup am facing problem with lower version of yealink phone and
grandstreams . Whenever 200 ok came in freeswitch changing IP phone ip to
asterisk PBX ip.It is showing messages in FS cli  switch_rtp.c "*Auto
changing IP PHONE IP to ASTERISK PBX IP *" . So no audio is passing to PBX
from from IP phone. Still IP phone able to get the incoming calls. Both PBX
and IP are in same network. Hopping the earliest reply .

Thanks And Regards

Andrew Paul
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From perazim at inbox.lt  Wed Apr 25 12:40:00 2012
From: perazim at inbox.lt (Perazim)
Date: Wed, 25 Apr 2012 11:40:00 +0300
Subject: [Freeswitch-users] question regarding rpm installation
Message-ID: <1335343200.2721.74.camel@travelmate.perazim.net>

I am a new user to freeswitch.

I have followed the wiki instructions for installing from rpms on a
fresh centos 6.2 i686 system with updates. I saw the note that the rpms
were beta but there should also be no problems.

This is the install command I used:

yum install freeswitch-config-vanilla

and it completed without errors however when I connected ekiga softphone to it, most of the test numbers failed.

My question is: Is this rpm based installation out of date or full of errors? 

Should I be installing from git?

Being a new user, I need something I can install that is known to work out of the box for learning how freeswitch works.

Thanks,

Perazim




From martin.soerensen at kontron.com  Wed Apr 25 13:46:42 2012
From: martin.soerensen at kontron.com (Martin Soerensen)
Date: Wed, 25 Apr 2012 11:46:42 +0200
Subject: [Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP
Message-ID: <CA2979A35DC83A4199BA719FE1FCB5D43224E3F653@KTDK05.KontronDK.local>

Hi all. Hope that somebody can help my...

First.......... I'm new to freeswitch... so please be patient ;o)

I'm trying to use freeswitch as a SBC.. lync isn't playing nice with NAT... so I'm sending all my call through FS trying to rewrite SIP headers

I'm having problems with ext-rtp-ip and SDP in my INVITE send by freeswitch.... the external IP is not set in FROM and SDP  when I dial out via lync.. I can set FROM by setting <param name="from-domain" value="xxx.xxx.157.92"/> but SDP is still my internal IP address



My setup :

10.70.0.182                           10.70.0.59                       xxx.xxx.157.92                                yyy.yyy.65.50
  Lync Srv    <------------------> FS <---------------> Firewall Nat <----------------> ISP/SIP gateway

FreeSWITCH Version 1.1.beta1 (git-2a25c4f 2012-04-24 16-30-20 +0200)

SIP trace:
send 1078 bytes to udp/[yyy.yyy.65.50]:5060 at 07:50:03.294218:
   ------------------------------------------------------------------------
   INVITE sip:+4529802936 at yyy.yyy.65.50:5060 SIP/2.0
   Via: SIP/2.0/UDP xxx.xxx.157.92;rport;branch=z9hG4bK9FZjDgDrUyHvD
   Max-Forwards: 69
   From: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;tag=7vjX1tBj987XS
   To: <sip:+4529802936 at yyy.yyy.65.50:5060>
   Call-ID: 1b96e79a-094e-1230-a29b-000c29236c5d
   CSeq: 27334037 INVITE
   Contact: <sip:gw+gc-gateway at xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
   User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-2a25c4f 2012-04-24 16-30-20 +0200
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 201
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1335307501 1335307502 IN IP4 10.70.0.59
   s=FreeSWITCH
   c=IN IP4 10.70.0.59
   t=0 0
   m=audio 32702 RTP/AVP 0 8 3 101 13
  a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------


sofia status profile external
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  10.70.0.59
Ext-RTP-IP              xxx.xxx.157.92
SIP-IP                  10.70.0.59
Ext-SIP-IP              xxx.xxx.157.92
URL                     sip:mod_sofia at xxx.xxx.157.92:5060
BIND-URL                sip:mod_sofia at xxx.xxx.157.92:5060;maddr=10.70.0.59
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               2
FAILED-CALLS-OUT        2
REGISTRATIONS           0


sofia status gateway gc-gateway
=================================================================================================
Name            gc-gateway
Profile         external
Scheme          Digest
Realm           yyy.yyy.65.50:5060
Username        User
Password        yes
>From            <sip:User at yyy.yyy.65.50:5060>
Contact         <sip:gw+gc-gateway at 1xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
Exten           User
To              sip:User at yyy.yyy.65.50:5060
Proxy           sip:yyy.yyy.65.50:5060
Context         public
Expires         3600
Freq            3600
Ping            1335339925
PingFreq        15
PingState       -1/1/1
State           NOREG
Status          UP
CallsIN         0
CallsOUT        0
FailedCallsIN   0
FailedCallsOUT  0
=================================================================================================

External profle:
<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="apply-nat-acl" value="rfc1918"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

Gateway
<?xml version="1.0"?>
<include>
      <gateway name="gc-gateway">
        <param name="realm" value="yyy.yyy.65.50:5060"/>
        <param name="username" value="User"/>
        <param name="password" value="Password"/>
        <param name="register" value="false"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="ping" value="15"/>
      </gateway>
</include>


Best regards
Martin


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From msc at freeswitch.org  Wed Apr 25 19:26:28 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 Apr 2012 08:26:28 -0700
Subject: [Freeswitch-users] Reminder: FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxWQqJuq3yPXXiNCEkft1CfSqYB50k+vNUjMMNGaef_8gA@mail.gmail.com>

Just reminding everyone of our call today:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25

We will be discussing the execution of dialplan apps and APIs using the
"execute_on" and "api_on" channel variables. We'll also have a lively
discussion about why shouldn't do billing stuff inline and why people do it
anyway. :)

See you at 1PM Eastern, 10AM Pacific

-Michael
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From Adam.Lappe at qsc.de  Wed Apr 25 19:30:36 2012
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 25 Apr 2012 17:30:36 +0200
Subject: [Freeswitch-users] park bridged call legs
In-Reply-To: <1FFF97C269757C458224B7C895F35F15099375@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F15099375@cantor.std.visionutv.se>
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D8BF8E454@CGNMSG01.oc.qsc.de>

Thank you Peter!
This is what i was looking for ;-)

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Peter Olsson
Gesendet: Mittwoch, 25. April 2012 15:23
An: 'FreeSWITCH Users Help'
Betreff: Re: [Freeswitch-users] park bridged call legs

Try this, I?m not sure if it works though.. :)

uuid_transfer <uuid> -both park inline

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Lappe, Adam
Skickat: den 25 april 2012 15:03
Till: 'FreeSWITCH Users Help'
?mne: Re: [Freeswitch-users] park bridged call legs

I guess I use the wrong syntax.
uuid_transfer <uuid> -both park
isn?t working. Any advice?

Thanks,
Adam

________________________________
Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:%5bmailto:freeswitch-users-bounces at lists.freeswitch.org%5d> Im Auftrag von Peter Olsson
Gesendet: Mittwoch, 25. April 2012 11:43
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] park bridged call legs

uuid_transfer should work for this. It can handle the flag ?-both?, which can transfer both legs somewhere.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:%5bmailto:freeswitch-users-bounces at lists.freeswitch.org%5d> F?r Lappe, Adam
Skickat: den 25 april 2012 11:22
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] park bridged call legs

Hi all,

is there a way to park two call legs after I bridged them?
When I use:
uuid_bridge <uuid_1> <uuid_2>
and then
uuid_park <uuid_1>
Freeswitch kills <uuid_2> with NORMAL CLEARING.

I tried to use park_after_bridge but this is not useful imo
since it parks every normal call when any1 hangs up.
Any ideas?

Thanks,
Adam

!DSPAM:4f97f4cc32761721410779!
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From anton.jugatsu at gmail.com  Wed Apr 25 19:32:56 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Wed, 25 Apr 2012 19:32:56 +0400
Subject: [Freeswitch-users] One way Audio
In-Reply-To: <CAMX4_Qkz3e1AU4pyUgdgz8nnorEG6wo-1yjakTk+cAsZdhw+9A@mail.gmail.com>
References: <CAMX4_Qkz3e1AU4pyUgdgz8nnorEG6wo-1yjakTk+cAsZdhw+9A@mail.gmail.com>
Message-ID: <CANBhyPBHGiF1Yd0hWS9owrpK_ZBf3y+5mrgsKZN6+xSuDOgH9A@mail.gmail.com>

Without any siptraces and logs or dialplan snippets it would be extremely
difficult to debug this one-way audio. Please, post this one. For that
purpose fs_logger (http://wiki.freeswitch.org/wiki/Fs_logger.pl) comes very
handy.

25 ?????? 2012 ?. 16:14 ???????????? Andrew Paul
<andrew.paul85 at gmail.com>???????:

> Hai,
> I have a freeswitch setup  in that one asterisk pbx registered. For this
> setup am facing problem with lower version of yealink phone and
> grandstreams . Whenever 200 ok came in freeswitch changing IP phone ip to
> asterisk PBX ip.It is showing messages in FS cli  switch_rtp.c "*Auto
> changing IP PHONE IP to ASTERISK PBX IP *" . So no audio is passing to
> PBX from from IP phone. Still IP phone able to get the incoming calls. Both
> PBX and IP are in same network. Hopping the earliest reply .
>
> Thanks And Regards
>
> Andrew Paul
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From adam.kelloway at newpace.ca  Wed Apr 25 20:01:53 2012
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Wed, 25 Apr 2012 13:01:53 -0300
Subject: [Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP
In-Reply-To: <CA2979A35DC83A4199BA719FE1FCB5D43224E3F653@KTDK05.KontronDK.local>
References: <CA2979A35DC83A4199BA719FE1FCB5D43224E3F653@KTDK05.KontronDK.local>
Message-ID: <4F981FF1.7040004@newpace.ca>

So this INVITE originates from your Lync Srv? When FreeSWITCH is used as 
a server, I believe it would detect that it doesn't need to use 
ext-rtp-ip because the INVITE originated inside the network. I have 
never used FreeSWITCH as a proxy, however. Have you checked out the 
wiki? http://wiki.freeswitch.org/wiki/Special:Search?search=SBC&go=Go

On 25/04/2012 6:46 AM, Martin Soerensen wrote:
>
> Hi all. Hope that somebody can help my...
>
> First.......... I'm new to freeswitch... so please be patient ;o)
>
> I'm trying to use freeswitch as a SBC.. lync isn't playing nice with 
> NAT... so I'm sending all my call through FS trying to rewrite SIP headers
>
> I'm having problems with ext-rtp-ip and SDP in my INVITE send by 
> freeswitch.... the external IP is not set in FROM and SDP  when I dial 
> out via lync.. I can set FROM by setting <param name="from-domain" 
> value="xxx.xxx.157.92"/> but SDP is still my internal IP address
>
> My setup :
>
> 10.70.0.182                           10.70.0.59                 
>       xxx.xxx.157.92                                yyy.yyy.65.50
>
>   Lync Srv <------------------> FS <---------------> Firewall Nat 
> <----------------> ISP/SIP gateway
>
> FreeSWITCH Version 1.1.beta1 (git-2a25c4f 2012-04-24 16-30-20 +0200)
>
> SIP trace:
>
> send 1078 bytes to udp/[yyy.yyy.65.50]:5060 at 07:50:03.294218:
>
>    
> ------------------------------------------------------------------------
>
>    INVITE sip:+4529802936 at yyy.yyy.65.50:5060 SIP/2.0
>
>    Via: SIP/2.0/UDP xxx.xxx.157.92;rport;branch=z9hG4bK9FZjDgDrUyHvD
>
>    Max-Forwards: 69
>
>    From: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;tag=7vjX1tBj987XS
>
>    To: <sip:+4529802936 at yyy.yyy.65.50:5060>
>
>    Call-ID: 1b96e79a-094e-1230-a29b-000c29236c5d
>
>    CSeq: 27334037 INVITE
>
>    Contact: 
> <sip:gw+gc-gateway at xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
>
>    User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-2a25c4f 2012-04-24 
> 16-30-20 +0200
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, 
> REGISTER, REFER, NOTIFY
>
>    Supported: timer, precondition, path, replaces
>
>    Allow-Events: talk, hold, refer
>
>    Content-Type: application/sdp
>
>    Content-Disposition: session
>
>    Content-Length: 201
>
>    X-FS-Support: update_display,send_info
>
>    Remote-Party-ID: "Martin Soerensen" 
> <sip:+4578772660 at 10.70.0.59>;party=calling;screen=yes;privacy=off
>
>    v=0
>
>    o=FreeSWITCH 1335307501 1335307502 IN IP4 10.70.0.59
>
>    s=FreeSWITCH
>
>    c=IN IP4 10.70.0.59
>
>    t=0 0
>
>    m=audio 32702 RTP/AVP 0 8 3 101 13
>
>   a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>    a=ptime:20
>
>    
> ------------------------------------------------------------------------
>
> sofia status profile external
>
> =================================================================================================
>
> Name                    external
>
> Domain Name             N/A
>
> Auto-NAT                false
>
> DBName                  sofia_reg_external
>
> Pres Hosts
>
> Dialplan                XML
>
> Context                 public
>
> Challenge Realm         auto_to
>
> RTP-IP                  10.70.0.59
>
> Ext-RTP-IP              xxx.xxx.157.92
>
> SIP-IP                  10.70.0.59
>
> Ext-SIP-IP              xxx.xxx.157.92
>
> URL sip:mod_sofia at xxx.xxx.157.92:5060
>
> BIND-URL sip:mod_sofia at xxx.xxx.157.92:5060;maddr=10.70.0.59
>
> HOLD-MUSIC              local_stream://moh
>
> OUTBOUND-PROXY          N/A
>
> CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
>
> CODECS OUT              PCMU,PCMA,GSM
>
> TEL-EVENT               101
>
> DTMF-MODE               rfc2833
>
> CNG                     13
>
> SESSION-TO              0
>
> MAX-DIALOG              0
>
> NOMEDIA                 false
>
> LATE-NEG                false
>
> PROXY-MEDIA             false
>
> AGGRESSIVENAT           false
>
> STUN-ENABLED            true
>
> STUN-AUTO-DISABLE       false
>
> CALLS-IN                0
>
> FAILED-CALLS-IN         0
>
> CALLS-OUT               2
>
> FAILED-CALLS-OUT        2
>
> REGISTRATIONS           0
>
> sofia status gateway gc-gateway
>
> =================================================================================================
>
> Name            gc-gateway
>
> Profile         external
>
> Scheme          Digest
>
> Realm           yyy.yyy.65.50:5060
>
> Username        User
>
> Password        yes
>
> From <sip:User at yyy.yyy.65.50:5060>
>
> Contact 
> <sip:gw+gc-gateway at 1xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
>
> Exten           User
>
> To sip:User at yyy.yyy.65.50:5060
>
> Proxy sip:yyy.yyy.65.50:5060
>
> Context         public
>
> Expires         3600
>
> Freq            3600
>
> Ping            1335339925
>
> PingFreq        15
>
> PingState       -1/1/1
>
> State           NOREG
>
> Status          UP
>
> CallsIN         0
>
> CallsOUT        0
>
> FailedCallsIN   0
>
> FailedCallsOUT  0
>
> =================================================================================================
>
> External profle:
>
> <profile name="external">
>
> <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
>
> <!-- This profile is only for outbound registrations to providers -->
>
> <gateways>
>
> <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>
> </gateways>
>
> <domains>
>
> <domain name="all" alias="false" parse="true"/>
>
> </domains>
>
> <settings>
>
> <param name="debug" value="0"/>
>
> <param name="sip-trace" value="no"/>
>
> <param name="sip-capture" value="no"/>
>
> <param name="rfc2833-pt" value="101"/>
>
> <param name="sip-port" value="$${external_sip_port}"/>
>
> <param name="dialplan" value="XML"/>
>
> <param name="context" value="public"/>
>
> <param name="dtmf-duration" value="2000"/>
>
> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>
> <param name="hold-music" value="$${hold_music}"/>
>
> <param name="rtp-timer-name" value="soft"/>
>
> <param name="local-network-acl" value="localnet.auto"/>
>
> <param name="manage-presence" value="false"/>
>
> <!--<param name="aggressive-nat-detection" value="true"/>-->
>
> <param name="apply-nat-acl" value="rfc1918"/>
>
> <param name="inbound-codec-negotiation" value="generous"/>
>
> <param name="nonce-ttl" value="60"/>
>
> <param name="auth-calls" value="false"/>
>
> <param name="rtp-ip" value="$${local_ip_v4}"/>
>
> <param name="sip-ip" value="$${local_ip_v4}"/>
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
> <param name="rtp-timeout-sec" value="300"/>
>
> <param name="rtp-hold-timeout-sec" value="1800"/>
>
> </settings>
>
> </profile>
>
> Gateway
>
> <?xml version="1.0"?>
>
> <include>
>
> <gateway name="gc-gateway">
>
> <param name="realm" value="yyy.yyy.65.50:5060"/>
>
> <param name="username" value="User"/>
>
> <param name="password" value="Password"/>
>
> <param name="register" value="false"/>
>
> <param name="caller-id-in-from" value="true"/>
>
> <param name="ping" value="15"/>
>
> </gateway>
>
> </include>
>
> Best regards
>
> Martin
>

-- 
Adam
-- 
NewPace Logo 	
	
	
	Adam Kelloway

	Software Engineer, NewPace
phone 	+1 (902) 406--8375  x1031
email 	Adam.Kelloway at NewPace.com <mailto:Adam.Kelloway at newpace.com>
aim <aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn 
<msnim:chat?contact=Adam.Kelloway at newpace.com> 	Adam.Kelloway 
<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

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From cmrienzo at gmail.com  Wed Apr 25 21:21:36 2012
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 25 Apr 2012 13:21:36 -0400
Subject: [Freeswitch-users] play_and_detect_speech errors
In-Reply-To: <CAGKPtBs=NABydjue-gxLNj6-G79qU5Q=+AZFXJ3m-tY=Y9UATw@mail.gmail.com>
References: <CAGKPtBs=NABydjue-gxLNj6-G79qU5Q=+AZFXJ3m-tY=Y9UATw@mail.gmail.com>
Message-ID: <CAD-qRkj6cbWXCifz6f5B6s72jJdjP+tBo-6jCzuFbOCwFJWWoA@mail.gmail.com>

Open a jira ticket and include the complete call trace.


On Wed, Apr 25, 2012 at 4:10 AM, Anita Hall <anita.hall at simmortel.com>wrote:

> Hi
>
> I was using play_and_detect_speech with Nuance Speech Server 5 and it
> generally works great! Much thanks to Chris Rienzo for this cool App :)
>
> There are 2 cases where I need help to understand what is happening.
>
> 1. In general my SIP connection between mod_unimrcp and Nuance works fine
> with all the calls answered. But in 3 repeated cases, the SIP connection
> was not established as a result of which the call got hungup.
>
> Is there any graceful way in play_and_detect_speech (like some channel
> variable set) which can be used to find if the SIP call has been answered
> by Nuance or not.
>
> Like in this case, I got
>
> 2012-04-20 13:20:47.955516 [DEBUG] apt_consumer_task.c:97 () Wait for
> Messages [MRCP Client]
>   7430 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
> () Receive SIP Event [nua_i_state] Status 0 INVITE sent
>   7431 2012-04-20 13:20:47.955516 [NOTICE]
> mrcp_sofiasip_client_agent.c:522 () SIP Call State 0x7f4f04188270 [calling]
>   7432 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
> () Receive SIP Event [nua_r_invite] Status 503 Service Unavailable
>   7433 2012-04-20 13:20:47.955516 [INFO] mrcp_sofiasip_client_agent.c:582
> () Receive SIP Event [nua_i_state] Status 503 Service Unavailable
>   7434 2012-04-20 13:20:47.955516 [NOTICE]
> mrcp_sofiasip_client_agent.c:522 () SIP Call State 0x7f4f04188270
> [terminated]
>   7435 2012-04-20 13:20:47.955516 [DEBUG] apt_task.c:278 () Process
> Message [MRCP Client] [1;4]
>   7436 2012-04-20 13:20:47.955516 [INFO] mrcp_client_session.c:535 (ASR-0)
> Raise App Response 0x7f4f04188270 <new> [2] FAILURE [2]
>   7437 2012-04-20 13:20:47.955516 [ERR] mod_unimrcp.c:1860 (ASR-0)
> RECOGNIZER channel error!
>   7438 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1866 Terminating
> MRCP session
>   7439 2012-04-20 13:20:47.955516 [DEBUG] mod_unimrcp.c:1541 (ASR-0)
> CLOSED ==> ERROR
>
>
>
>
>
>
> 2. After performing a recognition with a low confidence score, the call
> got disconnected but FreeSWITCH could not detect the call disconnection and
> repeatedly gave the following debug message in the log till it flooded
> hundreds of log files.
>
> switch_core_session.c:2102 FreeTDM/1:2/4948608 Channel is hungup and
> application play_and_detect_speech does not have the zombie_exec flag.
>
> During this "show calls" on fs_cli will crash fs_cli
>
> I had to shutdown freeswitch.
>
>
>
>
>
>
> regards,
> Anita
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From martin.soerensen at kontron.com  Wed Apr 25 22:20:03 2012
From: martin.soerensen at kontron.com (Martin Soerensen)
Date: Wed, 25 Apr 2012 20:20:03 +0200
Subject: [Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP
In-Reply-To: <4F981FF1.7040004@newpace.ca>
References: <CA2979A35DC83A4199BA719FE1FCB5D43224E3F653@KTDK05.KontronDK.local>
	<4F981FF1.7040004@newpace.ca>
Message-ID: <CA2979A35DC83A4199BA719FE1FCB5D43224E3F817@KTDK05.KontronDK.local>

I have a call coming from lync to FS... then I bridge the call to my gateway... it's the INVIT on the gateway to my ISP/VOIP provider where I have at private IP address in the INVIT... I'm trying to set external IP of my firewall... SIP is working fine.. but not media

Regards Martin


From: Adam Kelloway [mailto:adam.kelloway at newpace.ca]
Sent: 25. april 2012 18:02
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] INVITE ext-rtp-ip not set in FROM and SDP

So this INVITE originates from your Lync Srv? When FreeSWITCH is used as a server, I believe it would detect that it doesn't need to use ext-rtp-ip because the INVITE originated inside the network. I have never used FreeSWITCH as a proxy, however. Have you checked out the wiki? http://wiki.freeswitch.org/wiki/Special:Search?search=SBC&go=Go

On 25/04/2012 6:46 AM, Martin Soerensen wrote:
Hi all. Hope that somebody can help my...

First.......... I'm new to freeswitch... so please be patient ;o)

I'm trying to use freeswitch as a SBC.. lync isn't playing nice with NAT... so I'm sending all my call through FS trying to rewrite SIP headers

I'm having problems with ext-rtp-ip and SDP in my INVITE send by freeswitch.... the external IP is not set in FROM and SDP  when I dial out via lync.. I can set FROM by setting <param name="from-domain" value="xxx.xxx.157.92"/> but SDP is still my internal IP address



My setup :

10.70.0.182                           10.70.0.59                       xxx.xxx.157.92                                yyy.yyy.65.50
  Lync Srv    <------------------> FS <---------------> Firewall Nat <----------------> ISP/SIP gateway

FreeSWITCH Version 1.1.beta1 (git-2a25c4f 2012-04-24 16-30-20 +0200)

SIP trace:
send 1078 bytes to udp/[yyy.yyy.65.50]:5060 at 07:50:03.294218:
   ------------------------------------------------------------------------
   INVITE sip:+4529802936 at yyy.yyy.65.50:5060 SIP/2.0
   Via: SIP/2.0/UDP xxx.xxx.157.92;rport;branch=z9hG4bK9FZjDgDrUyHvD
   Max-Forwards: 69
   From: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;tag=7vjX1tBj987XS
   To: <sip:+4529802936 at yyy.yyy.65.50:5060>
   Call-ID: 1b96e79a-094e-1230-a29b-000c29236c5d
   CSeq: 27334037 INVITE
   Contact: <sip:gw+gc-gateway at xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
   User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-2a25c4f 2012-04-24 16-30-20 +0200
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 201
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Martin Soerensen" <sip:+4578772660 at 10.70.0.59>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1335307501 1335307502 IN IP4 10.70.0.59
   s=FreeSWITCH
   c=IN IP4 10.70.0.59
   t=0 0
   m=audio 32702 RTP/AVP 0 8 3 101 13
  a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------


sofia status profile external
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  10.70.0.59
Ext-RTP-IP              xxx.xxx.157.92
SIP-IP                  10.70.0.59
Ext-SIP-IP              xxx.xxx.157.92
URL                     sip:mod_sofia at xxx.xxx.157.92:5060
BIND-URL                sip:mod_sofia at xxx.xxx.157.92:5060;maddr=10.70.0.59
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               2
FAILED-CALLS-OUT        2
REGISTRATIONS           0


sofia status gateway gc-gateway
=================================================================================================
Name            gc-gateway
Profile         external
Scheme          Digest
Realm           yyy.yyy.65.50:5060
Username        User
Password        yes
>From            <sip:User at yyy.yyy.65.50:5060>
Contact         <sip:gw+gc-gateway at 1xxx.xxx.157.92:5060;transport=udp;gw=gc-gateway>
Exten           User
To              sip:User at yyy.yyy.65.50:5060
Proxy           sip:yyy.yyy.65.50:5060
Context         public
Expires         3600
Freq            3600
Ping            1335339925
PingFreq        15
PingState       -1/1/1
State           NOREG
Status          UP
CallsIN         0
CallsOUT        0
FailedCallsIN   0
FailedCallsOUT  0
=================================================================================================

External profle:
<profile name="external">
  <!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="sip-capture" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="apply-nat-acl" value="rfc1918"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

Gateway
<?xml version="1.0"?>
<include>
      <gateway name="gc-gateway">
        <param name="realm" value="yyy.yyy.65.50:5060"/>
        <param name="username" value="User"/>
        <param name="password" value="Password"/>
        <param name="register" value="false"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="ping" value="15"/>
      </gateway>
</include>


Best regards
Martin



--
Adam
--
[cid:image001.png at 01CD2320.CC18B7B0]

Adam Kelloway


Software Engineer, NewPace

phone

+1 (902) 406-8375  x1031

email

Adam.Kelloway at NewPace.com<mailto:Adam.Kelloway at newpace.com>

aim<aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn<msnim:chat?contact=Adam.Kelloway at newpace.com>

Adam.Kelloway<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca


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From d.campbell at ampersand.com  Thu Apr 26 00:11:36 2012
From: d.campbell at ampersand.com (Doug Campbell)
Date: Wed, 25 Apr 2012 16:11:36 -0400
Subject: [Freeswitch-users] help understanding crash
Message-ID: <4F985A78.5040609@ampersand.com>

We've been testing freeswitch in a high-volume application (100-200 simultaneous calls).
After running for about 1.5 hours successfully, it crashed.
Thankfully, it left a core.
Can anyone provide insight into this crash trace?

Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'.
Program terminated with signal 6, Aborted.
(gdb) bt
#0  0x0000003e4c432885 in raise () from /lib64/libc.so.6
#1  0x0000003e4c434065 in abort () from /lib64/libc.so.6
#2  0x0000003e4c42b9fe in __assert_fail_base () from /lib64/libc.so.6
#3  0x0000003e4c42bac0 in __assert_fail () from /lib64/libc.so.6
#4  0x00007fd2abb0e6bf in nua_destroy (nua=0x7fd29400c8a0) at nua.c:224
#5  0x00007fd2aba94bb4 in sofia_profile_thread_run (thread=<value optimized out>, obj=0x7fd294016bb0) at sofia.c:2085
#6  0x0000003e4cc077f1 in start_thread () from /lib64/libpthread.so.0
#7  0x0000003e4c4e592d in clone () from /lib64/libc.so.6
(gdb)

Any insight much appreciated,
Doug


From mthakershi at gmail.com  Thu Apr 26 00:15:15 2012
From: mthakershi at gmail.com (Malay Thakershi)
Date: Wed, 25 Apr 2012 15:15:15 -0500
Subject: [Freeswitch-users] Meaning of originate_timeout
Message-ID: <CAKHK4ZPmM6mttBr0aMSDVWccgN+pz5UrZn8d4Pd+2qqmOw-rgg@mail.gmail.com>

Hello,

When doing an originate, I set originate_timeout=45

Does it mean the called phone will ring for maximum 45 seconds?

Thanks.
Malay
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From anthony.minessale at gmail.com  Thu Apr 26 00:26:30 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 25 Apr 2012 15:26:30 -0500
Subject: [Freeswitch-users] help understanding crash
In-Reply-To: <4F985A78.5040609@ampersand.com>
References: <4F985A78.5040609@ampersand.com>
Message-ID: <CAKbxfG8tQzJO=hsA0=EWD_QCHPr-x8G0mKV=YWO1Tr+HMHd-xw@mail.gmail.com>

report bugs to http://jira.freeswitch.org

Reproduce with latest GIT HEAD and collect all relevant info.

Looking at this backtrace it looks like you are either unloading
mod_sofia or restarting the profile or something because its killing
the nua object.

You might be triggering the auto ip-change detection.
Make sure you disable auto-restart param in sofia.conf.xml and any sip profiles.

On Wed, Apr 25, 2012 at 3:11 PM, Doug Campbell <d.campbell at ampersand.com> wrote:
> We've been testing freeswitch in a high-volume application (100-200 simultaneous calls).
> After running for about 1.5 hours successfully, it crashed.
> Thankfully, it left a core.
> Can anyone provide insight into this crash trace?
>
> Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'.
> Program terminated with signal 6, Aborted.
> (gdb) bt
> #0 ?0x0000003e4c432885 in raise () from /lib64/libc.so.6
> #1 ?0x0000003e4c434065 in abort () from /lib64/libc.so.6
> #2 ?0x0000003e4c42b9fe in __assert_fail_base () from /lib64/libc.so.6
> #3 ?0x0000003e4c42bac0 in __assert_fail () from /lib64/libc.so.6
> #4 ?0x00007fd2abb0e6bf in nua_destroy (nua=0x7fd29400c8a0) at nua.c:224
> #5 ?0x00007fd2aba94bb4 in sofia_profile_thread_run (thread=<value optimized out>, obj=0x7fd294016bb0) at sofia.c:2085
> #6 ?0x0000003e4cc077f1 in start_thread () from /lib64/libpthread.so.0
> #7 ?0x0000003e4c4e592d in clone () from /lib64/libc.so.6
> (gdb)
>
> Any insight much appreciated,
> Doug
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From bernard.david.murphy at gmail.com  Wed Apr 25 23:26:40 2012
From: bernard.david.murphy at gmail.com (Bernard Murphy)
Date: Wed, 25 Apr 2012 20:26:40 +0100
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
Message-ID: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>

I have Freeswicth with a trunk to a Cisco SIP-ISDN gateway and am making a
call from a SIP device to an IVR out on the PSTN, i.e. SIP -> FREESWITCH ->
CISCO ->PSTN.  The Codec is PCMU throughout and while the call succeeds and
Freeswitch recognizes the DTMF and appears to generate tones in the Bleg
but they do not seem to be acceptable to the Cisco Gateway. The same call
without freeswitch in the middle i.e SIP -> CISCO -> PSTN works absolutely
fine.

I have already tried <action application="set"
data="rtp_manual_rtp_bugs=cisco_skip_mark_bit_2833"/> to no avail

A freeswitch log and wireshark trace are attached. Can anyone spot why this
is the case ?

I already admit that this is Freeswitch version 1.0.6 (rather old) and the
reason for that is that SIP Notify for MWI indication doesn't work with an
Avaya 9630 on anything later but do not have time to fix that right now.

Thanks in advance.
Murph
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From yehavi.bourvine at gmail.com  Thu Apr 26 09:24:36 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 26 Apr 2012 08:24:36 +0300
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
In-Reply-To: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
References: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
Message-ID: <CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>

Cisco is problematic with DTMF. Try using INFO instead of RFC-2833. You
even might need different encoding in each direction...

I tried finding the right config with Cisco's support and failed. I had to
replace the Cisco gateway with Audiocodes.

                         _Yehavi:

2012/4/25 Bernard Murphy <bernard.david.murphy at gmail.com>

> I have Freeswicth with a trunk to a Cisco SIP-ISDN gateway and am making a
> call from a SIP device to an IVR out on the PSTN, i.e. SIP -> FREESWITCH ->
> CISCO ->PSTN.  The Codec is PCMU throughout and while the call succeeds and
> Freeswitch recognizes the DTMF and appears to generate tones in the Bleg
> but they do not seem to be acceptable to the Cisco Gateway. The same call
> without freeswitch in the middle i.e SIP -> CISCO -> PSTN works absolutely
> fine.
>
> I have already tried <action application="set"
> data="rtp_manual_rtp_bugs=cisco_skip_mark_bit_2833"/> to no avail
>
> A freeswitch log and wireshark trace are attached. Can anyone spot why
> this is the case ?
>
> I already admit that this is Freeswitch version 1.0.6 (rather old) and the
> reason for that is that SIP Notify for MWI indication doesn't work with an
> Avaya 9630 on anything later but do not have time to fix that right now.
>
> Thanks in advance.
> Murph
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From francisco.cabrita at gmail.com  Thu Apr 26 03:53:47 2012
From: francisco.cabrita at gmail.com (Francisco Cabrita)
Date: Thu, 26 Apr 2012 00:53:47 +0100
Subject: [Freeswitch-users] OT: Weekly Conf Call 404
Message-ID: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>

Hi,

I'm trying to fetch the torrent file for today's conf call but I'm getting
a 404.

http://wiki.freeswitch.org/wiki/Weekly_Conference_Call

Can someone follow up this.

thanks in advance.

Francisco



-- 
Change is the essential process of all existence.

-- Spock, "Let That Be Your Last Battlefield", stardate 5730.2
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From msc at freeswitch.org  Thu Apr 26 09:37:27 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 25 Apr 2012 22:37:27 -0700
Subject: [Freeswitch-users] OT: Weekly Conf Call 404
In-Reply-To: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>
References: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>
Message-ID: <CAKzWOxVM+CgJErDhw8MSstk3RqEVK2ngJ+jSX7BxxdjH8rS1Ow@mail.gmail.com>

Sorry about that - I forgot to launch the torrent builder script. I just
launched it so the torrents should be ready any second now.

-MC

On Wed, Apr 25, 2012 at 4:53 PM, Francisco Cabrita <
francisco.cabrita at gmail.com> wrote:

> Hi,
>
> I'm trying to fetch the torrent file for today's conf call but I'm getting
> a 404.
>
> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call
>
> Can someone follow up this.
>
> thanks in advance.
>
> Francisco
>
>
>
> --
> Change is the essential process of all existence.
>
> -- Spock, "Let That Be Your Last Battlefield", stardate 5730.2
>
>
>
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From bernard.david.murphy at gmail.com  Thu Apr 26 10:16:50 2012
From: bernard.david.murphy at gmail.com (bernard.david.murphy at gmail.com)
Date: Thu, 26 Apr 2012 06:16:50 +0000
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
In-Reply-To: <CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>
References: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
	<CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>
Message-ID: <624329671-1335421011-cardhu_decombobulator_blackberry.rim.net-1496610603-@b28.c12.bise7.blackberry>

Info does work on outbound leg as you suggest but cisco will not generate info for an inbound leg as it considers it an obselete standard. So it would have to be different encoding in and out which is just messy.
Sent from my BlackBerry? wireless device

-----Original Message-----
From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 26 Apr 2012 08:24:36 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From william.king at quentustech.com  Thu Apr 26 11:11:45 2012
From: william.king at quentustech.com (William King)
Date: Thu, 26 Apr 2012 00:11:45 -0700
Subject: [Freeswitch-users] question regarding rpm installation
In-Reply-To: <1335343200.2721.74.camel@travelmate.perazim.net>
References: <1335343200.2721.74.camel@travelmate.perazim.net>
Message-ID: <4F98F531.7040908@quentustech.com>

You should probably check the configuration files that the vanilla
configs include. Last I checked they are very slimmed down versions of
the default configs. Most of those test extensions might not exist.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/25/2012 01:40 AM, Perazim wrote:
> I am a new user to freeswitch.
>
> I have followed the wiki instructions for installing from rpms on a
> fresh centos 6.2 i686 system with updates. I saw the note that the rpms
> were beta but there should also be no problems.
>
> This is the install command I used:
>
> yum install freeswitch-config-vanilla
>
> and it completed without errors however when I connected ekiga softphone to it, most of the test numbers failed.
>
> My question is: Is this rpm based installation out of date or full of errors? 
>
> Should I be installing from git?
>
> Being a new user, I need something I can install that is known to work out of the box for learning how freeswitch works.
>
> Thanks,
>
> Perazim
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From yehavi.bourvine at gmail.com  Thu Apr 26 13:20:56 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 26 Apr 2012 12:20:56 +0300
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
In-Reply-To: <624329671-1335421011-cardhu_decombobulator_blackberry.rim.net-1496610603-@b28.c12.bise7.blackberry>
References: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
	<CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>
	<624329671-1335421011-cardhu_decombobulator_blackberry.rim.net-1496610603-@b28.c12.bise7.blackberry>
Message-ID: <CAAtWKeGRvPy=uUHfMRhi97UREz-HbmgJ+RKFwCz7nqGf_BbsRA@mail.gmail.com>

On the other way you'll have to use RFC-2833...

I'll look later for backup configs I've made for my Cisco gateway and send
the relevant parts.

                    __Yehavi:

2012/4/26 <bernard.david.murphy at gmail.com>

> Info does work on outbound leg as you suggest but cisco will not generate
> info for an inbound leg as it considers it an obselete standard. So it
> would have to be different encoding in and out which is just messy.
> Sent from my BlackBerry? wireless device
>
> -----Original Message-----
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Thu, 26 Apr 2012 08:24:36
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] DTMF on BLeg not recognized by Cisco
> Gateway
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From rico-freeswitch at ricozome.net  Thu Apr 26 13:21:41 2012
From: rico-freeswitch at ricozome.net (rico-freeswitch at ricozome.net)
Date: Thu, 26 Apr 2012 11:21:41 +0200
Subject: [Freeswitch-users] SIP invite contact header... again
Message-ID: <4F9913A5.505@ricozome.net>

Hi list,

I have some analogic phones (to be axact two Siemens Gigaset DECT
phones) connected to my FS server through a RTC-to-SIP gateway.

It is working well, except for announcing the caller phone number. After
some investigations, it appear the SIP gateway expect the caller number
into the "Contact" SIP header like this :

Contact: <sip:105002 at 46.28.168.58:5060>

I found a lot of documentation related to manipulating SIP header on
gateways, but in that particular case, I need to alter SIP contact
header from and to a particular extension, and most of call don't even
transit through a SIP gateway as they are handled into the same SIP profile.

I guess this is trivial, but I'm still a newbie on IPBX and FS, and
explanations I found on the wiki are too obsure for me...

Thanks for your help,

-- 
Eric


From francisco.cabrita at gmail.com  Thu Apr 26 14:10:38 2012
From: francisco.cabrita at gmail.com (Francisco Cabrita)
Date: Thu, 26 Apr 2012 11:10:38 +0100
Subject: [Freeswitch-users] OT: Weekly Conf Call 404
In-Reply-To: <CAKzWOxVM+CgJErDhw8MSstk3RqEVK2ngJ+jSX7BxxdjH8rS1Ow@mail.gmail.com>
References: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>
	<CAKzWOxVM+CgJErDhw8MSstk3RqEVK2ngJ+jSX7BxxdjH8rS1Ow@mail.gmail.com>
Message-ID: <CAAyAuB+Uv9MMihQddpH7Hyda4=CMTPmBX1=_375gEFfbashVWw@mail.gmail.com>

On Thu, Apr 26, 2012 at 6:37 AM, Michael Collins <msc at freeswitch.org> wrote:

> Sorry about that - I forgot to launch the torrent builder script. I just
> launched it so the torrents should be ready any second now.


No problem. Many thanks :)

Francisco


>
> -MC
>
>
> On Wed, Apr 25, 2012 at 4:53 PM, Francisco Cabrita <
> francisco.cabrita at gmail.com> wrote:
>
>> Hi,
>>
>> I'm trying to fetch the torrent file for today's conf call but I'm
>> getting a 404.
>>
>> http://wiki.freeswitch.org/wiki/Weekly_Conference_Call
>>
>> Can someone follow up this.
>>
>> thanks in advance.
>>
>> Francisco
>>
>>
>>
>> --
>> Change is the essential process of all existence.
>>
>> -- Spock, "Let That Be Your Last Battlefield", stardate 5730.2
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Change is the essential process of all existence.

-- Spock, "Let That Be Your Last Battlefield", stardate 5730.2
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From pkelly at gmail.com  Thu Apr 26 15:06:02 2012
From: pkelly at gmail.com (Pete Kelly)
Date: Thu, 26 Apr 2012 12:06:02 +0100
Subject: [Freeswitch-users] Receiving "early media" during call
Message-ID: <CAEWH9M_1eDMZ8uJwyxEnJQ8oS+5N2Yq+p52fM8gdr2UUrLJxpw@mail.gmail.com>

Hi

I have a scenario where there is an established call (originated from a
freeswitch bridge to a downstream UAC)

The downstream UA then initiates a re-invite and freeswitch accepts and
keeps sending the 200OK with SDP as normal.

However the downstream UA is also sending a 183 with sdp to freeswitch at
this time, and it looks like freeswitch is ignoring this media.

Is it possible to get freeswitch to accept this 183 media during a reinvite
transaction?

Pete
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From tpe at actimizer.com  Thu Apr 26 16:02:18 2012
From: tpe at actimizer.com (Tor Petterson)
Date: Thu, 26 Apr 2012 14:02:18 +0200
Subject: [Freeswitch-users] Sending RTP while parked
Message-ID: <CADnUQS_6cAk+Di0ga36-dhRE3s4fX1m2PnPbxF7QV8WrAGpDbg@mail.gmail.com>

Hi

I am having the problem that my telephony provider is dropping my
calls when they have been parked for five minutes because Freeswitch
doesn't send any RTP packets while a call is parked.
Is there any way to make Freeswitch send RTP packet while parked?
Either periodically or all the time?

-- 
Tor Petterson

tpe at actimizer.com

Tobaksvejen 25, 2. tv. - 2860 S?borg
Telephone: +45 39 55 05 32
www.actimizer.com


From bernard.david.murphy at gmail.com  Thu Apr 26 16:15:37 2012
From: bernard.david.murphy at gmail.com (bernard.david.murphy at gmail.com)
Date: Thu, 26 Apr 2012 12:15:37 +0000
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
In-Reply-To: <CAAtWKeGRvPy=uUHfMRhi97UREz-HbmgJ+RKFwCz7nqGf_BbsRA@mail.gmail.com>
References: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
	<CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>
	<624329671-1335421011-cardhu_decombobulator_blackberry.rim.net-1496610603-@b28.c12.bise7.blackberry>
	<CAAtWKeGRvPy=uUHfMRhi97UREz-HbmgJ+RKFwCz7nqGf_BbsRA@mail.gmail.com>
Message-ID: <990357526-1335442537-cardhu_decombobulator_blackberry.rim.net-951413854-@b28.c12.bise7.blackberry>

To do that I'd have to have another sip profile for inbound call legs on a different port I suppose ?

Sent from my BlackBerry? wireless device

-----Original Message-----
From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 26 Apr 2012 12:20:56 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From peter.olsson at visionutveckling.se  Thu Apr 26 16:42:03 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 26 Apr 2012 12:42:03 +0000
Subject: [Freeswitch-users] Sending RTP while parked
Message-ID: <1FFF97C269757C458224B7C895F35F1509A86F@cantor.std.visionutv.se>

send_silence_when_idle should be enough.

http://wiki.freeswitch.org/wiki/Variable_send_silence_when_idle

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tor Petterson
Skickat: den 26 april 2012 14:02
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] Sending RTP while parked

Hi

I am having the problem that my telephony provider is dropping my
calls when they have been parked for five minutes because Freeswitch
doesn't send any RTP packets while a call is parked.
Is there any way to make Freeswitch send RTP packet while parked?
Either periodically or all the time?

-- 
Tor Petterson

tpe at actimizer.com

Tobaksvejen 25, 2. tv. - 2860 S?borg
Telephone: +45 39 55 05 32
www.actimizer.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4f9937fd32761218312289!



From william.suffill at gmail.com  Thu Apr 26 19:27:46 2012
From: william.suffill at gmail.com (William Suffill)
Date: Thu, 26 Apr 2012 11:27:46 -0400
Subject: [Freeswitch-users] OT: Weekly Conf Call 404
In-Reply-To: <CAAyAuB+Uv9MMihQddpH7Hyda4=CMTPmBX1=_375gEFfbashVWw@mail.gmail.com>
References: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>
	<CAKzWOxVM+CgJErDhw8MSstk3RqEVK2ngJ+jSX7BxxdjH8rS1Ow@mail.gmail.com>
	<CAAyAuB+Uv9MMihQddpH7Hyda4=CMTPmBX1=_375gEFfbashVWw@mail.gmail.com>
Message-ID: <CAF-ZooSo3PfVe0QRKZQ05ziC2LDOMgfJH=QMvNOMuHSquEZBcg@mail.gmail.com>

4-11 torrent 404's as well


From jlang at singo1.com  Thu Apr 26 19:14:55 2012
From: jlang at singo1.com (Jesse Lang)
Date: Thu, 26 Apr 2012 10:14:55 -0500
Subject: [Freeswitch-users] OT - Communication System Integrator position
 available with Singo Solution
Message-ID: <4F99666F.6080606@singo1.com>

Previously posted on the Freeswitch-Biz list:
-----------------------------------------------------------------------------------
Singo Solution provides hosted software contact management solutions for 
a number of industries, including call centers, utilities, telecom 
providers, education, and medical. We are based in the Phoenix area, and 
have presence throughout the US as well as internationally.

Singo has an opening for a full-time employee position as a 
Communication System Integrator. Employee must live in the Phoenix area. 
Responsibilities include traveling to client sites (in and around 
Phoenix) to integrate their telephony and communication systems with 
products and solutions offered by Singo and our strategic partners. 
Benefits include medical and dental plans, and the ability to work from 
home when not traveling to client sites.

Required skills include: Fluent knowledge of Linux as a server platform, 
Extensive knowledge of Freeswitch/FreeTDM, Basic knowledge of 
Asterisk/DAHDI, along with basic office applications and e-mail experience.

Interested parties should send their resume to hr at singo1.com. Thank you!


From msc at freeswitch.org  Thu Apr 26 20:24:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 Apr 2012 09:24:36 -0700
Subject: [Freeswitch-users] OT: Weekly Conf Call 404
In-Reply-To: <CAF-ZooSo3PfVe0QRKZQ05ziC2LDOMgfJH=QMvNOMuHSquEZBcg@mail.gmail.com>
References: <CAAyAuBJ+trMQr-TJcn3rNzshusp1+PpXfj-BVdfDiD_OzDvi9g@mail.gmail.com>
	<CAKzWOxVM+CgJErDhw8MSstk3RqEVK2ngJ+jSX7BxxdjH8rS1Ow@mail.gmail.com>
	<CAAyAuB+Uv9MMihQddpH7Hyda4=CMTPmBX1=_375gEFfbashVWw@mail.gmail.com>
	<CAF-ZooSo3PfVe0QRKZQ05ziC2LDOMgfJH=QMvNOMuHSquEZBcg@mail.gmail.com>
Message-ID: <CAKzWOxWsnOT7YD1yTFAQB2JXaNTEAu0t_k7tGMkisjNm2tT5Dg@mail.gmail.com>

Okay, give it a few minutes and try again.
-MC

On Thu, Apr 26, 2012 at 8:27 AM, William Suffill
<william.suffill at gmail.com>wrote:

> 4-11 torrent 404's as well
>
>
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From gabe at gundy.org  Thu Apr 26 23:21:52 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 26 Apr 2012 13:21:52 -0600
Subject: [Freeswitch-users] mod_callcenter and prompts to members
In-Reply-To: <E1SN1rJ-00069x-V7.joohny-mail-ru@f215.mail.ru>
References: <E1SN1rJ-00069x-V7.joohny-mail-ru@f215.mail.ru>
Message-ID: <CAMwZ-tKOLyQrROKXGp4uSkk5hzDPSeyqqt4Hj_HiGLA+YCoDsw@mail.gmail.com>

On Wed, Apr 25, 2012 at 7:02 AM, ??????? <joohny at mail.ru> wrote:
> How can I gues It? Or there is another way to solve this issue?
> I want to prompt something like: you are third in a queue.

Sorry, I'm no help. I thought this was built in, but I guess not. Have
you figured anything out?


Gabe


From kerem.erciyes at gmail.com  Fri Apr 27 01:05:53 2012
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Fri, 27 Apr 2012 00:05:53 +0300
Subject: [Freeswitch-users] FreeSWITCH and Sangoma - Robbed Bit Channel
 Associated Signaling (CAS)
In-Reply-To: <333789DE5C38474EB3A478A538F4EBAB0A333F0025@prod-exch01.corp.vseinc.com>
References: <333789DE5C38474EB3A478A538F4EBAB0A333F0025@prod-exch01.corp.vseinc.com>
Message-ID: <CAJ6wjQFLzx5GR4UfWdjGBU=28CZ7R4HK807WGvMM-LSu=iF7EA@mail.gmail.com>

FS logs, FreeTDM Config and Wanpipe Config files might be helpful.

http://pastebin.freeswitch.org

PS. I only did CAS on Sangoma -> Rhino CB24 with CAS, was not on
FreeSwitch, but had to do some wanrouter config to set it up.




On Tue, Apr 24, 2012 at 8:56 PM, <bedgar at vseinc.com> wrote:

> Are there any tricks to getting Robbed Bit Channel Associated Signaling
> (CAS) working properly with FS (legacy 1.06 nor latest git seem to show
> desired results) and Sangoma boards.  I am using model A102, REV 1.1.  DTMF
> does not seem to be recognized in the call stream so IVR does not respond.
> SIP calls work fine.****
>
> ** **
>
> Brian C. Edgar, Jr.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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From curriegrad2004 at gmail.com  Fri Apr 27 03:05:25 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 26 Apr 2012 16:05:25 -0700
Subject: [Freeswitch-users] FreeSWITCH and Sangoma - Robbed Bit Channel
 Associated Signaling (CAS)
In-Reply-To: <CAJ6wjQFLzx5GR4UfWdjGBU=28CZ7R4HK807WGvMM-LSu=iF7EA@mail.gmail.com>
References: <333789DE5C38474EB3A478A538F4EBAB0A333F0025@prod-exch01.corp.vseinc.com>
	<CAJ6wjQFLzx5GR4UfWdjGBU=28CZ7R4HK807WGvMM-LSu=iF7EA@mail.gmail.com>
Message-ID: <CAMWiMy_F-mKi0Uj+Ui02HUiRVujgUWu-2q0XHmE=pq-yYqsb3g@mail.gmail.com>

Robbed bit signalling only denotes the state of the call (i.e ringing
or busy, etc) . I'd say you need an in-band DTMF detector for this to
work. mod_spandsp already provides this functionality of what you're
looking for. The same goes for the opposite when you are calling from
SIP-> TDM (meaning you have to have a way to generate in-band DTMF
tones from the RFC 2833 signals).

On Thu, Apr 26, 2012 at 2:05 PM, Kerem Erciyes <kerem.erciyes at gmail.com> wrote:
> FS logs, FreeTDM Config and Wanpipe Config files might be helpful.
>
> http://pastebin.freeswitch.org
>
> PS. I only did CAS on Sangoma -> Rhino CB24 with CAS, was not on FreeSwitch,
> but had to do some wanrouter config to set it up.
>
>
>
>
> On Tue, Apr 24, 2012 at 8:56 PM, <bedgar at vseinc.com> wrote:
>>
>> Are there any tricks to getting Robbed Bit Channel Associated Signaling
>> (CAS) working properly with FS (legacy 1.06 nor latest git seem to show
>> desired results) and Sangoma boards.? I am using model A102, REV 1.1.? DTMF
>> does not seem to be recognized in the call stream so IVR does not respond.
>> SIP calls work fine.
>>
>>
>>
>> Brian C. Edgar, Jr.
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kerem Erciyes - Sistem Danismani
> http://keremerciyes.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rml at tollfreeforwarding.com  Fri Apr 27 04:01:30 2012
From: rml at tollfreeforwarding.com (RaviRaj Mulasa)
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curl
Message-ID: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>

Hi

Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH.

>From Linux command prompt

SUCCESS
curl -F file=@/tmp/log/callcontrol.log http://192.168.9.152/upload


>From fs_cli

curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>
curl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>

All the above results in

<html>
<head><title>415 Unsupported Media Type</title></head>
<body bgcolor="white">
<center><h1>415 Unsupported Media Type</h1></center>
<hr><center>nginx/1.2.0</center>
</body>
</html>

FreeSWITCH version
FreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000)

Thanks
RaviRaj.
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From joohny at mail.ru  Fri Apr 27 09:17:00 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Fri, 27 Apr 2012 09:17:00 +0400
Subject: [Freeswitch-users] =?utf-8?q?autoanswer_by_re-INVITE?=
Message-ID: <E1SNdYC-00067m-H9.joohny-mail-ru@f125.mail.ru>

Hello.
There is no way to do this?


----------------------------------
Hi!
I can send a new call with autoanswer by
<action application="export" data="sip_auto_answer=true"/>
It works well.
But I need some different feature
I want to "re-INVITE" existing call with autoanswer option, how can I do this?
Or may be somebody knows, how can I make Cisco SPA508G to answer the call with any command from FreeSWITCH. It will be very usefull with headset to answer calls by pressing some buttons with keybord or mouse on PC.
From Russia with Love :)
Evginey
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From joohny at mail.ru  Fri Apr 27 09:20:48 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Fri, 27 Apr 2012 09:20:48 +0400
Subject: [Freeswitch-users]
	=?utf-8?q?mod=5Fcallcenter_and_prompts_to_memb?= =?utf-8?q?ers?=
Message-ID: <E1SNdbs-00035g-8n.joohny-mail-ru@f75.mail.ru>

Hello.

Does anybody do such prompts to queue? Or just listening to music and simple phrases?

Thank you.
--------------------------------------
Hello.
I try to make prompting script for queues.?
So I want to know who will be first, second ... last in
freeswitch at 8.8.8.8@default> callcenter_config queue list members 3001
queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state
3001|single_box|cd025909-0751-4f46-9208-34302eb2d34d|392b4df4-45fa-4b1b-8e42-d15117edce5e|0000000000||1335357680|1335357685|0|0|0|5|0|||Waiting
3001|single_box|88006efb-f80b-47fd-a4d6-1c711483d3d3|92e77fde-f2ce-45d7-885b-7475b3ef9b48|0000000000||1335357683|1335357687|0|0|0|4|0|||Waiting
3001|single_box|c6c74d6a-092c-49a4-9242-abba925c1523|69db31f3-786d-4f2a-ae2b-3f5954c0e4c4|0000000000||1335357684|1335357689|0|0|0|5|0|||Waiting
3001|single_box|a2509a89-5717-44cc-8382-c5718f1c9813|8f692a80-6680-4eff-8692-9f71f1843508|0000000000||1335357685|1335357690|0|0|0|5|0|||Waiting
3001|single_box|38dd8346-97cf-4d3a-9e8b-773103e4657f|382e0341-cc24-48de-84c2-d27583bd8115|0000000000||1335357687|1335357692|0|0|0|5|0|||Waiting
3001|single_box|beba7fc8-2eb8-4ff6-bcb3-c08599e0dc19|56e2c2f9-c458-4df6-b3eb-dae4b9ae0a5a|0000000000||1335357688|1335357693|0|0|0|5|0|||Waiting
3001|single_box|c4552675-2be1-4e12-8ca5-a9d20a421be2|8d47431f-c7da-4f4a-9074-ebbbee6a0710|0000000000||1335357690|1335357694|0|0|0|4|0|||Waiting
3001|single_box|89b0ada0-44fb-4f65-98ec-182589fe3fe9|f7263ffd-05ff-4184-a6fe-2aaede34692f|0000000000||1335357691|1335357696|0|0|0|5|0|||Waiting
3001|single_box|72b2bd2e-87fb-4912-a190-fda98ee8cef8|9cede828-db7c-48fb-85f4-6cbd2069c264|0000000000||1335357692|1335357697|0|0|0|5|0|||Waiting
+OK

How can I gues It? Or there is another way to solve this issue?
I want to prompt something like: you are third in a queue.


From Russia with Love :)
Evginey?
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From brian at freeswitch.org  Fri Apr 27 09:21:22 2012
From: brian at freeswitch.org (Brian West)
Date: Fri, 27 Apr 2012 00:21:22 -0500
Subject: [Freeswitch-users] autoanswer by re-INVITE
In-Reply-To: <E1SNdYC-00067m-H9.joohny-mail-ru@f125.mail.ru>
References: <E1SNdYC-00067m-H9.joohny-mail-ru@f125.mail.ru>
Message-ID: <1513239168368441670@unknownmsgid>

you don't:

enable the talk package and use:

uuid_phone_event <uuid> talk

/b

Sent from my iPad

On Apr 27, 2012, at 12:19 AM, "???????" <joohny at mail.ru> wrote:

Hello.
There is no way to do this?


----------------------------------
Hi!

I can send a new call with autoanswer by
<action application="export" data="sip_auto_answer=true"/>
It works well.
But I need some different feature
I want to "re-INVITE" existing call with autoanswer option, how can I do
this?

Or may be somebody knows, how can I make Cisco SPA508G to answer the call
with any command from FreeSWITCH. It will be very usefull with headset to
answer calls by pressing some buttons with keybord or mouse on PC.

>From Russia with Love :)
Evginey

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From joohny at mail.ru  Fri Apr 27 09:46:53 2012
From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=)
Date: Fri, 27 Apr 2012 09:46:53 +0400
Subject: [Freeswitch-users] =?utf-8?q?mod=5Fcallcenter_and_prompts_to_memb?=
	=?utf-8?q?ers?=
Message-ID: <E1SNe17-0004jW-Pk.joohny-mail-ru@f295.mail.ru>

Hello.

Thank you for answer. 

There is no way documented in wiki. So I thought that members are in sorted list and answers comes to the first one, then the second... But No! Also There is field "base_score". This is time in seconds - but many members can have same value if there are a lot of calls.
I have no ideas about that. :(
Evginey?

On Wed, Apr 25, 2012 at 7:02 AM, ??????? <joohny at mail.ru> wrote:
> How can I gues It? Or there is another way to solve this issue?
> I want to prompt something like: you are third in a queue.

Sorry, I'm no help. I thought this was built in, but I guess not. Have
you figured anything out?


Gabe

Hello.

Does anybody do such prompts to queue? Or just listening to music and simple phrases?

Thank you.
--------------------------------------
Hello.
I try to make prompting script for queues.?
So I want to know who will be first, second ... last in
freeswitch at 8.8.8.8@default> callcenter_config queue list members 3001
queue|system|uuid|session_uuid|cid_number|cid_name|system_epoch|joined_epoch|rejoined_epoch|bridge_epoch|abandoned_epoch|base_score|skill_score|serving_agent|serving_system|state
3001|single_box|cd025909-0751-4f46-9208-34302eb2d34d|392b4df4-45fa-4b1b-8e42-d15117edce5e|0000000000||1335357680|1335357685|0|0|0|5|0|||Waiting
3001|single_box|88006efb-f80b-47fd-a4d6-1c711483d3d3|92e77fde-f2ce-45d7-885b-7475b3ef9b48|0000000000||1335357683|1335357687|0|0|0|4|0|||Waiting
3001|single_box|c6c74d6a-092c-49a4-9242-abba925c1523|69db31f3-786d-4f2a-ae2b-3f5954c0e4c4|0000000000||1335357684|1335357689|0|0|0|5|0|||Waiting
3001|single_box|a2509a89-5717-44cc-8382-c5718f1c9813|8f692a80-6680-4eff-8692-9f71f1843508|0000000000||1335357685|1335357690|0|0|0|5|0|||Waiting
3001|single_box|38dd8346-97cf-4d3a-9e8b-773103e4657f|382e0341-cc24-48de-84c2-d27583bd8115|0000000000||1335357687|1335357692|0|0|0|5|0|||Waiting
3001|single_box|beba7fc8-2eb8-4ff6-bcb3-c08599e0dc19|56e2c2f9-c458-4df6-b3eb-dae4b9ae0a5a|0000000000||1335357688|1335357693|0|0|0|5|0|||Waiting
3001|single_box|c4552675-2be1-4e12-8ca5-a9d20a421be2|8d47431f-c7da-4f4a-9074-ebbbee6a0710|0000000000||1335357690|1335357694|0|0|0|4|0|||Waiting
3001|single_box|89b0ada0-44fb-4f65-98ec-182589fe3fe9|f7263ffd-05ff-4184-a6fe-2aaede34692f|0000000000||1335357691|1335357696|0|0|0|5|0|||Waiting
3001|single_box|72b2bd2e-87fb-4912-a190-fda98ee8cef8|9cede828-db7c-48fb-85f4-6cbd2069c264|0000000000||1335357692|1335357697|0|0|0|5|0|||Waiting
+OK

How can I gues It? Or there is another way to solve this issue?
I want to prompt something like: you are third in a queue.


From Russia with Love :)
Evginey?
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From bdfoster at endigotech.com  Fri Apr 27 10:02:13 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 27 Apr 2012 02:02:13 -0400
Subject: [Freeswitch-users] autoanswer by re-INVITE
In-Reply-To: <1513239168368441670@unknownmsgid>
References: <E1SNdYC-00067m-H9.joohny-mail-ru@f125.mail.ru>
	<1513239168368441670@unknownmsgid>
Message-ID: <CAKV-jFwJuvSUVHfeiuwTfrndJ3gT70nZhC77hDh4cS7zz-YK2Q@mail.gmail.com>

Can we get uuid_phone_event documented? I don't see anything on the wiki on
this. There could be some good stuff in there like 'talk'.

-BDF
On Apr 27, 2012 1:22 AM, "Brian West" <brian at freeswitch.org> wrote:

> you don't:
>
> enable the talk package and use:
>
> uuid_phone_event <uuid> talk
>
> /b
>
> Sent from my iPad
>
> On Apr 27, 2012, at 12:19 AM, "???????" <joohny at mail.ru> wrote:
>
> Hello.
> There is no way to do this?
>
>
> ----------------------------------
> Hi!
>
> I can send a new call with autoanswer by
> <action application="export" data="sip_auto_answer=true"/>
> It works well.
> But I need some different feature
> I want to "re-INVITE" existing call with autoanswer option, how can I do
> this?
>
> Or may be somebody knows, how can I make Cisco SPA508G to answer the call
> with any command from FreeSWITCH. It will be very usefull with headset to
> answer calls by pressing some buttons with keybord or mouse on PC.
>
> From Russia with Love :)
> Evginey
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Fri Apr 27 11:51:00 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Fri, 27 Apr 2012 11:51:00 +0400
Subject: [Freeswitch-users] mod_voicemail and non-numeric id
Message-ID: <CANBhyPAb7F9=iV5BQ7a5EcQct+q1Y5uo_MGhNBpvw5VjyzCG6A@mail.gmail.com>

Hello list.

I've decided to use non-numeric id's at users configuration.

<user id="001565229c95" number-alias="109" mailbox="109">

But when the call goes to voicemail app

<action application="bridge" data="loopback/app=voicemail:default
${domain_name} ${dialed_extension}"/>

switch_ivr_play_say.c:244 Handle say:[001565229c95] (ru:ru)
2012-04-27 11:39:54.452208 [INFO] mod_say_ru.c:721 ru_say!!!  001565229c95!
  say_opt.gender=0   say_opt.cases=0
2012-04-27 11:39:54.532157 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 16000hz 1 channels 20ms

The mod_say pronounces user id. How to force to pronounce number-alias?
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From yehavi.bourvine at gmail.com  Fri Apr 27 12:57:21 2012
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 27 Apr 2012 11:57:21 +0300
Subject: [Freeswitch-users] DTMF on BLeg not recognized by Cisco Gateway
In-Reply-To: <990357526-1335442537-cardhu_decombobulator_blackberry.rim.net-951413854-@b28.c12.bise7.blackberry>
References: <CAOte9J4bj7qut39khd28ahi024TPYubSWtZxzaZ2vmwkXATOsw@mail.gmail.com>
	<CAAtWKeHAgQ8sxgLmvJUZ+GGGS0DzKmeFf4zgenWp3H1jojSWLA@mail.gmail.com>
	<624329671-1335421011-cardhu_decombobulator_blackberry.rim.net-1496610603-@b28.c12.bise7.blackberry>
	<CAAtWKeGRvPy=uUHfMRhi97UREz-HbmgJ+RKFwCz7nqGf_BbsRA@mail.gmail.com>
	<990357526-1335442537-cardhu_decombobulator_blackberry.rim.net-951413854-@b28.c12.bise7.blackberry>
Message-ID: <CAAtWKeGzPRizVbTW7gofy6d1Em5EUV252q7rMhzMAUqYhthubA@mail.gmail.com>

Hi,

  Yes, you need two profiles - one for incoming and one for outgoing. I
enclose here the relevant fragments from my old config:

voice service voip
 qsig decode
 dtmf-interworking rtp-nte    (RFC-2833)
 srtp fallback
 allow-connections sip to sip
 fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback cisco
Not sure this is relevant:

interface Serial0/0/0:15
 no ip address
 encapsulation hdlc
 isdn switch-type primary-qsig
 isdn overlap-receiving
 isdn not-end-to-end 64
 isdn incoming-voice voice
 isdn supp-service name calling profile Network-Extension
operation-value-tag local
 isdn negotiate-bchan
 isdn outgoing ie facility
 isdn outgoing ie notify-indicator
 isdn outgoing display-ie
 isdn outgoing ie caller-number
 isdn outgoing ie called-number
 no cdp enable
This is for calls from PSTN to VoIP:

dial-peer voice 201 voip
 destination-pattern 40.....
 progress_ind setup enable 1
 progress_ind progress enable 8
 session protocol sipv2
 session target ipv4:10.1.1.1:5070  Note the special port - it lands on
a different profile
 session transport udp
 dtmf-relay sip-notify          Use INFO/NOTIFY instead of RFC-2833
 codec g711alaw
 fax-relay sg3-to-g3
 fax protocol t38 ls-redundancy 4 hs-redundancy 2 fallback cisco
 no vad

The VoIP to PSTN has nothing special:
dial-peer voice 1 pots
 destination-pattern 8T
 progress_ind setup enable 3
 progress_ind progress enable 8
 no digit-strip
 direct-inward-dial
 port 0/0/0:15

                   Regards, __Yehavi:

2012/4/26 <bernard.david.murphy at gmail.com>

> To do that I'd have to have another sip profile for inbound call legs on a
> different port I suppose ?
>
> Sent from my BlackBerry? wireless device
>
> -----Original Message-----
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Thu, 26 Apr 2012 12:20:56
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] DTMF on BLeg not recognized by Cisco
> Gateway
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From Rob.Moore at Aeriandi.com  Fri Apr 27 15:33:33 2012
From: Rob.Moore at Aeriandi.com (Rob Moore)
Date: Fri, 27 Apr 2012 11:33:33 +0000
Subject: [Freeswitch-users] Eaves Drop
Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local>

Hi All,

Hopefully looking for a quick answer here. We are using the Eavesdrop function to supply a "call barging" service for a call centre we are running through a web UI but we have a few problems around transfers where we park the call and switching the barge from one agent to another.

Basically I need some method of removing the media bug from the agents channel. I've searched the Wiki and done a few googles but I can't seem to find anything that we can call to remove a bug.

We're doing all of our control of freeswitch through lua so anything that we can call from the CLI would be great.

Anyone have any suggestions?

Thanks in advance!

Rob
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From rajkumar.kanniappan at sasken.com  Fri Apr 27 15:44:32 2012
From: rajkumar.kanniappan at sasken.com (rajprithiv89)
Date: Fri, 27 Apr 2012 04:44:32 -0700 (PDT)
Subject: [Freeswitch-users] Re ceiving only RTP packets
Message-ID: <33758451.post@talk.nabble.com>


Hi,

we are using csipsimple android voip application which uses Freeswitch as
SIP/Media server.

We established voip calls between two android clients(A, B) by enabling SRTP
in both the sides. 

Now the problem observed here is, when we captured the packet flow in
wireshark, the client which initiates the call(A) is sending SRTP packets
and the client which receives the call sends only RTP packets and the
FreeSwitch sends SRTP packets to both the clients.

We have tested FreeSwitch with other application, there also we ended up in
the same problem(one sided SRTP packets flow).
Also we have tested the same CSipSimple application with Asterisk server. It
works fine by sending SRTP packets from both the clients.


When i captured the Wireshark packets SDP contains:


>From A to Server:

m= audio 4006 RTP/SAVP 98 97 99



>From Server to B

m = audio 31562 RTP/SAVP 3 98 99 9
m = audio 31562 RTP/AVP 3 98 99 9
(sends two media line, one for SRTP and one for RTP)



>From B to Server(Negotiation done)

m = audio 4014 RTP/SAVP 3 96
m = audio 0 RTP/AVP 3 98 99 9   (This is to inform, B is not going to use
RTP by putting the port number as 0)


>From Server to A(Negotiation done)

m = audio 32338 RTP/SAVP 3 96



But in the end,
A is sending SRTP packets
Server is sending SRTP packets
B is sending only RTP packets (But in the negotiation RTP/AVP port was 0)


-- 
View this message in context: http://old.nabble.com/Receiving-only-RTP-packets-tp33758451p33758451.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From vipkilla at gmail.com  Fri Apr 27 17:55:15 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 27 Apr 2012 09:55:15 -0400
Subject: [Freeswitch-users] FS does not follow XML standard for '<' or '>'
Message-ID: <CAC-LwPODskVmBOUzdKBAjTVdRcjeUsYnSm8ZeKHoZuGbOnS2Dw@mail.gmail.com>

The XML standard says that > should be escaped as &gt; and < should be
escaped as &lt;
Unfortunately when using &lt; or &gt; in dialplan FS does not read
those as < or >
This is a problem when using some tools to generate XML such as perl's
XML::Writer.
Is there anyway the FS dialplan can modified to follow these XML standards?


From peter.olsson at visionutveckling.se  Fri Apr 27 18:11:57 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 27 Apr 2012 14:11:57 +0000
Subject: [Freeswitch-users] FS does not follow XML standard for '<' or
 '>'
Message-ID: <1FFF97C269757C458224B7C895F35F1509BDD5@cantor.std.visionutv.se>

In what case do you really need it? Please note that it's only for escaping strings inside <>-tags, and I can't really remember any case where that is needed in the FS config.

Can you provide an example generated file?

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Vik Killa
Skickat: den 27 april 2012 15:55
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] FS does not follow XML standard for '<' or '>'

The XML standard says that > should be escaped as &gt; and < should be escaped as &lt; Unfortunately when using &lt; or &gt; in dialplan FS does not read those as < or > This is a problem when using some tools to generate XML such as perl's XML::Writer.
Is there anyway the FS dialplan can modified to follow these XML standards?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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FreeSWITCH-users mailing list
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!DSPAM:4f9aa40532761819614895!



From avlubimov at gmail.com  Fri Apr 27 18:31:17 2012
From: avlubimov at gmail.com (Lubimov Alexey)
Date: Fri, 27 Apr 2012 18:31:17 +0400
Subject: [Freeswitch-users] wrong detection dtmf rfc2833
Message-ID: <4F9AADB5.9060200@gmail.com>

Good day!

I have problem with dtmf on incoming calls.

Internal dtmf work normal, but on incoming call i have many mistakes 
(111  or 121 instead 110, for example).

siptrace wrong detection attached, Abonent send 110, but freeswitch 
detect 111.

in profile's config i set these options. (Im trying many variants, but 
no find solution)

<param name="dtmf-duration" value="2000"/>
<param name="dtmf-type" value="rfc2833"/>
<param name="rfc2833-pt" value="0"/>
<param name="liberal-dtmf" value="true"/>

Have you any ideas?



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From covici at ccs.covici.com  Fri Apr 27 18:37:54 2012
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 27 Apr 2012 10:37:54 -0400
Subject: [Freeswitch-users] FS does not follow XML standard for '<' or
	'>'
In-Reply-To: <1FFF97C269757C458224B7C895F35F1509BDD5@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F1509BDD5@cantor.std.visionutv.se>
Message-ID: <31233.1335537474@ccs.covici.com>

I have used it in my cdr-csv and it is working for me.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> In what case do you really need it? Please note that it's only for escaping strings inside <>-tags, and I can't really remember any case where that is needed in the FS config.
> 
> Can you provide an example generated file?
> 
> /Peter
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Vik Killa
> Skickat: den 27 april 2012 15:55
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS does not follow XML standard for '<' or '>'
> 
> The XML standard says that > should be escaped as &gt; and < should be escaped as &lt; Unfortunately when using &lt; or &gt; in dialplan FS does not read those as < or > This is a problem when using some tools to generate XML such as perl's XML::Writer.
> Is there anyway the FS dialplan can modified to follow these XML standards?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4f9aa40532761819614895!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From vipkilla at gmail.com  Fri Apr 27 18:49:02 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 27 Apr 2012 10:49:02 -0400
Subject: [Freeswitch-users] FS does not follow XML standard for '<' or
	'>'
In-Reply-To: <31233.1335537474@ccs.covici.com>
References: <1FFF97C269757C458224B7C895F35F1509BDD5@cantor.std.visionutv.se>
	<31233.1335537474@ccs.covici.com>
Message-ID: <CAC-LwPMkU_8wsr8D0TTWBSV2kf+cbkCc78y95bRGhzwk0GufzA@mail.gmail.com>

Example, this works in FS:
<condition field="${cond(${record_out}>0?YES:NO)}" expression="^YES$">

This should also work:
<condition field="${cond(${record_out} &gt ;0?YES:NO)}" expression="^YES$">


From vipkilla at gmail.com  Fri Apr 27 19:22:23 2012
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 27 Apr 2012 11:22:23 -0400
Subject: [Freeswitch-users] FS does not follow XML standard for '<' or
	'>'
In-Reply-To: <CAC-LwPMkU_8wsr8D0TTWBSV2kf+cbkCc78y95bRGhzwk0GufzA@mail.gmail.com>
References: <1FFF97C269757C458224B7C895F35F1509BDD5@cantor.std.visionutv.se>
	<31233.1335537474@ccs.covici.com>
	<CAC-LwPMkU_8wsr8D0TTWBSV2kf+cbkCc78y95bRGhzwk0GufzA@mail.gmail.com>
Message-ID: <CAC-LwPPc9NJ5u8Jan+K0i5JEW5vVrvrRTahssOAXCXx4xJU6ig@mail.gmail.com>

Apologies, FS does properly use &gt;
It was my XML_CURL app, that was generating &amp;gt;
Thanks!


From arkaha at hotbox.ru  Fri Apr 27 19:14:53 2012
From: arkaha at hotbox.ru (Limit)
Date: Fri, 27 Apr 2012 08:14:53 -0700 (PDT)
Subject: [Freeswitch-users] Attended transfer via ESL
Message-ID: <1335539693783-7506468.post@n2.nabble.com>

I'm trying to make an attended transfer through EventSocket, using sendmsg
with att_xfer command, but having problems with it.

ESL function:
execute("att_xfer", args, b_uuid);

It creates new calls as normal dialplan att_xfer, but problem occured when
one of three sides hangs up.
Sometimes it goes as planned, but more often all channels hangup
automatically.

att_xfer called from dialplan work as needed, looks like there is some
problem with EventSocket call specifically . Maybe I must set some variables
before to use it properly?

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7506468.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fs-list at communicatefreely.net  Fri Apr 27 20:06:15 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 27 Apr 2012 12:06:15 -0400
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <CAKV-jFydnHBYZkrPVxS8Py6yRijThc6NF_f54mEVpAUHhuEuSw@mail.gmail.com>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>	<CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>	<B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>
	<CAKV-jFydnHBYZkrPVxS8Py6yRijThc6NF_f54mEVpAUHhuEuSw@mail.gmail.com>
Message-ID: <4F9AC3F7.1000601@communicatefreely.net>

Hi Rick,

Freeswitch has no problem with this at all, but finding a phone that can 
do it may be the issue.

You would need to find a phone that can configure BLFs using a specified 
SIP domain, not just the SIP username.

For example, on an Aastra phone (it's what I know), you configure the 
SIP domain that is associated with a given registration.  When you 
create BLF keys, you only specify the username - the domain that is 
subscribed to is whatever the domain is for that registration.

Somehow, you need to map a button and corresponding subscription to each 
receptionist phone at their respective domain, and this may not be easy.

Off the top of my head, I don't know which phones can do this.  Polycom 
might be able to do it, as they seem to be better at making the full SIP 
URI accessible to the user.  I know it's possible to dial a SIP URI 
directly on a Polycom, whereas Aastra phones have to have it in the 
speed dial list.

Last I checked, there was some sort of option to only let people 
subscribe to their own domain, but I think it is off by default.

Another thing you will need to check, is the "force-subsription-domain" 
and "force-subscription-db-domain" (sic) in your SIP profile to make 
sure that the domain you subscribe to is what is actually used in the 
subscription.

-Tim

Brian Foster wrote:
>
>
> On Tue, Apr 24, 2012 at 12:43 AM, Rick Guyton 
> <Rick at claritycomputer.com <mailto:Rick at claritycomputer.com>> wrote:
>
>     To be clear, I?m in utter amazement of what freeswitch can do and
>     appreciate everyone who has contributed to making such an awesome
>     application.
>
>
> Welcome to the club ;-)
>  
>
>     So, when I can?t do something, I assume it is because of my own
>     stupidity. I had assumed that I needed to make a ?permissions?
>     change somewhere or add the users from other domains as an alias
>     or some sort. If it can?t be done, that makes sense. I meant no
>     disrespect to Tony or anyone else who has contributed. I would
>     just put them all in the same domain, but I must separate the
>     companies for mostly political reasons.
>
>
> Understandable.
>
> Depending on your setup, you might just be able to do this on the 
> phones themselves. I don't know if you use these, but you know the 
> little attendant consoles you can plug into phones like polycom and 
> aastra? Can you register to the lines you want to monitor with those? 
> I've never actually had one (I'd like to get one for testing) but I'm 
> fairly certain you can. That would also depend on your network setup. 
> If you've got all the companies on their own LAN's then you're screwed 
> unless freeswitch can be fully accessed on each of those LAN's. Might 
> be something to check out.
>
>     * *
>
> -BDF 
>
>      
>
>     *From:* freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
>     Of *Chris Chen
>     *Sent:* Monday, April 23, 2012 6:30 PM
>     *To:* FreeSWITCH Users Help
>     *Subject:* Re: [Freeswitch-users] BLF in Multi-tenant config
>
>      
>
>     Agree with BDF.
>
>      
>
>     By design, in a multi-tenant environment, 101 at companyA should only
>     have BLF status for any extension within companyA,  otherwise why
>     should you have multi-tenant setup, just have them all in the same
>     domain. We should really appreciate Tony crew having built the FS
>     to be really capable of providing multi-tenant functions without
>     too many tweaks. 
>
>      
>
>     Just my two cents.
>
>      
>
>     Chris
>
>      
>
>     On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster
>     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
>
>     I'm not sure that multi-tenant wouldn't support that. I think the
>     point of multi-tenant is to keep everything seperated. There might
>     be a hack for this, though. Nothing comes to mind for me.
>
>     -BDF
>
>     On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com
>     <mailto:Rick at claritycomputer.com>> wrote:
>
>         We have 3 different companies in this office and we all share
>         the same freeswitch server, I?ll call then CompanyA, CompanyB
>         and CompanyC. CompanyA, B and C all employ their own
>         receptionists, and they are all assigned extension 101 on in
>         their respective domains. However, the receptionists all work
>         very tightly together and basically share work between them.
>         So, I need to assign the BLF status of 101 at CompanyA,
>         101 at CompanyB and 101 at CompanyC to indicator lights across all
>         phones in the building. My problem is that users in CompanyA
>         can?t see the status of 101 at CompanyB or 101 at CompanyC. Users in
>         CompanyB can?t see 101 at CompanyA or 101 at CompanyC and so on?
>
>          
>
>         So, my question is, how do I allow users of one domain in a
>         multi-tenant configuration to see the status of other users in
>         a different domain?
>
>          
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>      
>
>      
>
>     ------------------------------------------------------------------------
>
>     If this email is spam, report it to www.OnlyMyEmail.com
>     <https://support.onlymyemail.com/view/report_spam/MTM3Mjc1OjE0MTI0MDQ4MjI6cmlja0BjbGFyaXR5Y29tcHV0ZXIuY29tOmRlbGl2ZXJlZA>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Brian D. Foster
> Endigo Computer LLC
> Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
> Phone: 317-800-7876
> Indianapolis, Indiana, USA
>
> This message contains confidential information and is intended for 
> those listed in the "To:", "CC:", and/or "BCC:" fields of the message 
> header. If you are not the intended recipient you are notified that 
> disclosing, copying, distributing or taking any action in reliance on 
> the contents of this information is strictly prohibited. E-mail 
> transmission cannot be guaranteed to be secure or error-free as 
> information could be intercepted, corrupted, lost, destroyed, arrive 
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>
> ------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>   



From fs-list at communicatefreely.net  Fri Apr 27 20:14:13 2012
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 27 Apr 2012 12:14:13 -0400
Subject: [Freeswitch-users] client change IP address while ringing
In-Reply-To: <BCCA7469823E440096A6D6BC76E07037@imsdeveloper>
References: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>	<CALn233qM80E8izqKwkyM9sqEg9sJ_2Np-HoJ8FHbZN6683TYKQ@mail.gmail.com>
	<BCCA7469823E440096A6D6BC76E07037@imsdeveloper>
Message-ID: <4F9AC5D5.3030509@communicatefreely.net>

That's an impossible situation -

How does FS know the new address?  The only way is if the phone 
registers.  FS won't even know where to send the next packet until the 
phone does another registration so it gets an update.

If you have a registration for both the WiFi and 3G address, you will 
have better luck, but I think FS will still maintain the contact info 
for a dialog that has already started.

If this is a problem you really need to solve, then you may need to set 
up some sort of VPN so that a common IP address can be reachable over 
both 3G and WiFi.  Even then, there is still going to be a period of 
instability when you transition, as the routing has to re-converge.

You also can't transition an active call between WiFi and 3G for the 
same reasons.  The only way to do that would be to park the call, 
re-register on the new network, then pick it up again.

Hope that's helpful.

-Tim


Mario Karakanovski wrote:
> Thank you Mitch,
>
> 	The client use TCP, but the problem is that it must change IP
> address, because it's switch between WiFi and 3G. So, there is no chance to
> force it using the same IP address.
>
> Regards,
> 	Mario
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
> Capper
> Sent: Tuesday, April 24, 2012 4:24 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] client change IP address while ringing
>
> Try changing the mobile client to register via TCP I would assume then
> the IP will not change as that would break the TCP connection:)
>
> ~Mitch
>
> Hi guys,
>
>  
>
>             I have an interesting situation.
>
> I have two mobile clients and connect them using bridge. It is working
> without problems, except when the called party changes its IP address
> between ringing and answer.
>
> Does anybody deal with such situation?
>
>  
>
> Regards,
>
>             Mario  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From chrisbware at yahoo.it  Fri Apr 27 20:25:26 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Fri, 27 Apr 2012 17:25:26 +0100 (BST)
Subject: [Freeswitch-users] Time routing issue
Message-ID: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>

Hi,

I'd like to set a channel variable and use it in a time condition like this:

<condition mon="${rmon}" >


with rmon=1-12 (just to be sure condition is verified).

I can't find out why it doesn't work.?

- Using?<action application="info"/> , I can see variable is set [1-12]
- I've tried also?<condition mon="${rmon}" ?inline="true">
- Using directly ?<condition mon="1-12" > it works

What's wrong? Help, please
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From erin.omeara at salmonbaytechnology.com  Fri Apr 27 20:25:43 2012
From: erin.omeara at salmonbaytechnology.com (Erin O'Meara)
Date: Fri, 27 Apr 2012 09:25:43 -0700
Subject: [Freeswitch-users] How do I...
Message-ID: <CAB4Oy0VnUZ=A7vGbsqdtRHuFHH6g298xQGC9=XuNFhm2o_7i5w@mail.gmail.com>

These are a few things I want to do with freeswitch (using FusionPBX), but
I'm not sure what the best/easiest way would be. Not so much looking
for explanations on how to solve the problems, more what direction I should
read/research to learn how to solve these problems with freeswitch. But if
you have seen one of the questions answered on the web I will gladly not
re-invent the wheel.

I realize the there are only two variables, time + money. If one of these
ideas is doable, but I'm going to have to spend weeks learning the in's and
out's please mention that :)

1. The same customer calls 3 times in five minutes, if they reach my
voicemail prompt and call back within 5 minutes redirect them straight to
voicemail.

Suggested in IRC to set a dialplan XML variable with a timestamp per
incoming callerid,  if they call back and timestamp is less then five
minutes send straight to voicemail. if this is posible, does not seam like
the best solution because the variables would never expire without some
expire script?

2. I have a hunt group with my desk phone extension, cellphone sip client
extension(only active on WiFi) and external forward to cellphone PSTN
number.
When I get a call, my sip phones ring instantly, and a few seconds later my
cellphone rings thru the PSTN.

Can I enable/disable the cellphone forward based on if my cellphone sip
client is registered?
For my huntgroup, if I have an active call send the call straight to
voicemail/busy message? using limit?

3. Can I have nested hunt groups (I have not tired), basically main number
rings extensions and my personal huntgroup. If my hunt group is busy, its
still rings the other phones before it does its fall back rule.

Regards,

206.905.9520
 http://salmonbaytechnology.com
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From avlubimov at gmail.com  Fri Apr 27 21:15:27 2012
From: avlubimov at gmail.com (=?KOI8-R?B?4czFy9PFyiDswMLJzc/X?=)
Date: Fri, 27 Apr 2012 21:15:27 +0400
Subject: [Freeswitch-users] wrong detection dtmf rfc2833
In-Reply-To: <4F9AADB5.9060200@gmail.com>
References: <4F9AADB5.9060200@gmail.com>
Message-ID: <CAA_857vSoH-1BC8QbsyD=XJ0F2nf30qwBvkw5OGBt=PvbMPiPQ@mail.gmail.com>

I think, this bug this bug back again.

http://jira.freeswitch.org/browse/FS-2604

In my case, dtmf recieved in rfc2833, but, enyway, i have dublicated dtmf
in ivr.


I do number 23, but freeswitch recieve 223.

EXECUTE sofia/external/79168030237 at 195.211.120.9:5061 ivr(neolant)
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-exit'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-sub'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-exec-app'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-play-sound'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-back'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665
switch_ivr_menu_stack_xml_add binding 'menu-top'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:796 building menu
'neolant'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:857 binding menu
action 'menu-exec-app' to '/^([0-9][0-9][0-9])$/'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu
neolant
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:67 No language
specified - Using [ru]
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[ivr/ivr-enter_ext_pound.wav] (ru:ru)
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:1309 Codec
Activated L16 at 8000hz 1 channels 20ms
2012-04-28 01:01:38.707451 [DEBUG] switch_ivr_async.c:2253 DTMF DETECTED:
[2][320]
2012-04-28 01:01:38.707451 [DEBUG] switch_ivr_play_say.c:1682 done playing
file /opt/freeswitch/sounds/ru/RU/elena/ivr/ivr-enter_ext_pound.wav
2012-04-28 01:01:38.967450 [DEBUG] switch_ivr_menu.c:348 waiting for 2/3
digits t/o 3000
2012-04-28 01:01:39.107457 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2:1600
2012-04-28 01:01:40.267453 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 3:1600
2012-04-28 01:01:40.267453 [DEBUG] switch_ivr_menu.c:395 digits '223'
2012-04-28 01:01:40.267453 [DEBUG] switch_ivr_menu.c:489 action regex [223]
[/^([0-9][0-9][0-9])$/] [2]




2012/4/27 Lubimov Alexey <avlubimov at gmail.com>

> Good day!
>
> I have problem with dtmf on incoming calls.
>
> Internal dtmf work normal, but on incoming call i have many mistakes (111
>  or 121 instead 110, for example).
>
> siptrace wrong detection attached, Abonent send 110, but freeswitch detect
> 111.
>
> in profile's config i set these options. (Im trying many variants, but no
> find solution)
>
> <param name="dtmf-duration" value="2000"/>
> <param name="dtmf-type" value="rfc2833"/>
> <param name="rfc2833-pt" value="0"/>
> <param name="liberal-dtmf" value="true"/>
>
> Have you any ideas?
>
>
>
>
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From jeff at jefflenk.com  Fri Apr 27 22:13:46 2012
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 27 Apr 2012 11:13:46 -0700 (PDT)
Subject: [Freeswitch-users] Time routing issue
In-Reply-To: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
References: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
Message-ID: <1335550426288-7507046.post@n2.nabble.com>

Are you setting the rmon variable in the same dialplan execution? If so you
will need to inline the set of the variable.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Time-routing-issue-tp7506728p7507046.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nasida at live.ru  Fri Apr 27 22:50:23 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 27 Apr 2012 22:50:23 +0400
Subject: [Freeswitch-users] wrong detection dtmf rfc2833
In-Reply-To: <CAA_857vSoH-1BC8QbsyD=XJ0F2nf30qwBvkw5OGBt=PvbMPiPQ@mail.gmail.com>
References: <4F9AADB5.9060200@gmail.com>,
	<CAA_857vSoH-1BC8QbsyD=XJ0F2nf30qwBvkw5OGBt=PvbMPiPQ@mail.gmail.com>
Message-ID: <SNT127-W16680B02C8CBF4363B6079D5250@phx.gbl>


it depend from your  incoming sip provider as well.did you try to detect dtmf  for incoming calls  in media stream (inband)  instead of using rfc2833 ? <param name="dtmf-type" value="none"/>  - on  incoming  sip profile<action application="start_dtmf" /> - in dialplan before bridge
Date: Fri, 27 Apr 2012 21:15:27 +0400
From: avlubimov at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] wrong detection dtmf rfc2833

I think, this bug this bug back again.

http://jira.freeswitch.org/browse/FS-2604

In my case, dtmf recieved in rfc2833, but, enyway, i have dublicated dtmf in ivr.



I do number 23, but freeswitch recieve 223.

EXECUTE sofia/external/79168030237 at 195.211.120.9:5061 ivr(neolant)
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exit'

2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-sub'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-exec-app'

2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-play-sound'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-back'

2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:665 switch_ivr_menu_stack_xml_add binding 'menu-top'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:796 building menu 'neolant'
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:857 binding menu action 'menu-exec-app' to '/^([0-9][0-9][0-9])$/'

2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_menu.c:433 Executing IVR menu neolant
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [ru]
2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[ivr/ivr-enter_ext_pound.wav] (ru:ru)

2012-04-28 01:01:33.987457 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16 at 8000hz 1 channels 20ms
2012-04-28 01:01:38.707451 [DEBUG] switch_ivr_async.c:2253 DTMF DETECTED: [2][320]
2012-04-28 01:01:38.707451 [DEBUG] switch_ivr_play_say.c:1682 done playing file /opt/freeswitch/sounds/ru/RU/elena/ivr/ivr-enter_ext_pound.wav

2012-04-28 01:01:38.967450 [DEBUG] switch_ivr_menu.c:348 waiting for 2/3 digits t/o 3000
2012-04-28 01:01:39.107457 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2:1600
2012-04-28 01:01:40.267453 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 3:1600

2012-04-28 01:01:40.267453 [DEBUG] switch_ivr_menu.c:395 digits '223'
2012-04-28 01:01:40.267453 [DEBUG] switch_ivr_menu.c:489 action regex [223] [/^([0-9][0-9][0-9])$/] [2]





2012/4/27 Lubimov Alexey <avlubimov at gmail.com>

Good day!



I have problem with dtmf on incoming calls.



Internal dtmf work normal, but on incoming call i have many mistakes (111  or 121 instead 110, for example).



siptrace wrong detection attached, Abonent send 110, but freeswitch detect 111.



in profile's config i set these options. (Im trying many variants, but no find solution)



<param name="dtmf-duration" value="2000"/>

<param name="dtmf-type" value="rfc2833"/>

<param name="rfc2833-pt" value="0"/>

<param name="liberal-dtmf" value="true"/>



Have you any ideas?










_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From nasida at live.ru  Fri Apr 27 22:56:52 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 27 Apr 2012 22:56:52 +0400
Subject: [Freeswitch-users] File upload - mod_curl
In-Reply-To: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>
Message-ID: <SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>


For example , you can to launch some lua script from FS dialplan and invoke linux command from lua via 
io.popen()http://wiki.freeswitch.org/wiki/Mod_lua#Run_a_shell_command 

From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curl

Hi  Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH. From Linux command prompt SUCCESScurl -F file=@/tmp/log/callcontrol.log http://192.168.9.152/upload  From fs_cli curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.logcurl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log All the above results in <html><head><title>415 Unsupported Media Type</title></head><body bgcolor="white"><center><h1>415 Unsupported Media Type</h1></center><hr><center>nginx/1.2.0</center></body></html> FreeSWITCH versionFreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000) ThanksRaviRaj.
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Fri Apr 27 23:00:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 27 Apr 2012 14:00:42 -0500
Subject: [Freeswitch-users] How do I...
In-Reply-To: <CAB4Oy0VnUZ=A7vGbsqdtRHuFHH6g298xQGC9=XuNFhm2o_7i5w@mail.gmail.com>
References: <CAB4Oy0VnUZ=A7vGbsqdtRHuFHH6g298xQGC9=XuNFhm2o_7i5w@mail.gmail.com>
Message-ID: <CAKbxfG8PmsD-cG2j6r=40yeLXNzzUyuvM7oj-Y32EKdDWxB8mQ@mail.gmail.com>

On Fri, Apr 27, 2012 at 11:25 AM, Erin O'Meara <
erin.omeara at salmonbaytechnology.com> wrote:

> These are a few things I want to do with freeswitch (using FusionPBX), but
> I'm not sure what the best/easiest way would be. Not so much looking
> for explanations on how to solve the problems, more what direction I should
> read/research to learn how to solve these problems with freeswitch. But if
> you have seen one of the questions answered on the web I will gladly not
> re-invent the wheel.
>
> I realize the there are only two variables, time + money. If one of these
> ideas is doable, but I'm going to have to spend weeks learning the in's and
> out's please mention that :)
>
> 1. The same customer calls 3 times in five minutes, if they reach my
> voicemail prompt and call back within 5 minutes redirect them straight to
> voicemail.
>
> Suggested in IRC to set a dialplan XML variable with a timestamp per
> incoming callerid,  if they call back and timestamp is less then five
> minutes send straight to voicemail. if this is posible, does not seam like
> the best solution because the variables would never expire without some
> expire script?
>
>
You probably want to develop this logic as an application using a database
and use either embedded lua script in an application or a xml_curl dialplan
so you can make the logic in a normal application and return the correct
dialplan based on the circumstances.




> 2. I have a hunt group with my desk phone extension, cellphone sip client
> extension(only active on WiFi) and external forward to cellphone PSTN
> number.
> When I get a call, my sip phones ring instantly, and a few seconds later
> my cellphone rings thru the PSTN.
>
> Can I enable/disable the cellphone forward based on if my cellphone sip
> client is registered?
> For my huntgroup, if I have an active call send the call straight to
> voicemail/busy message? using limit?
>
>
Also solved with dynamic dialplan (xml_curl) and logic on the caller id
data.




> 3. Can I have nested hunt groups (I have not tired), basically main number
> rings extensions and my personal huntgroup. If my hunt group is busy, its
> still rings the other phones before it does its fall back rule.
>
> Regards,
>

There are some pretty extensive documentation on dialing and groups.  Look
for the originate syntax and/or enterprise originate.




>
> 206.905.9520
>  http://salmonbaytechnology.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From rml at tollfreeforwarding.com  Fri Apr 27 23:18:07 2012
From: rml at tollfreeforwarding.com (RaviRaj Mulasa)
Date: Fri, 27 Apr 2012 19:18:07 +0000
Subject: [Freeswitch-users] File upload - mod_curl
In-Reply-To: <SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>
	<SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>
Message-ID: <8B94625BC339264DBA61E314BE9EC2CF2E9CC774@EXCH125.IFN.com>

In my use I am using outbound socket for FAX
After receiving a 'CUTOM - spandsp::rxfaxresult ' event , I need to call curl/http application  to upload the received fax from remote FreeSWITCH to storage.
I want to run the file upload process in background - bgapi curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>.

Regards
RaviRaj.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] File upload - mod_curl

For example , you can to launch some lua script from FS dialplan and invoke linux command from lua via  io.popen()
http://wiki.freeswitch.org/wiki/Mod_lua#Run_a_shell_command
________________________________
From: rml at tollfreeforwarding.com<mailto:rml at tollfreeforwarding.com>
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curl
Hi

Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH.

>From Linux command prompt

SUCCESS
curl -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log> http://192.168.9.152/upload


>From fs_cli

curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>
curl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>

All the above results in

<html>
<head><title>415 Unsupported Media Type</title></head>
<body bgcolor="white">
<center><h1>415 Unsupported Media Type</h1></center>
<hr><center>nginx/1.2.0</center>
</body>
</html>

FreeSWITCH version
FreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000)

Thanks
RaviRaj.

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org<mailto:consulting at freeswitch.org> http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at 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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From nasida at live.ru  Fri Apr 27 23:39:35 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Fri, 27 Apr 2012 23:39:35 +0400
Subject: [Freeswitch-users] File upload - mod_curl
In-Reply-To: <8B94625BC339264DBA61E314BE9EC2CF2E9CC774@EXCH125.IFN.com>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>,
	<SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>,
	<8B94625BC339264DBA61E314BE9EC2CF2E9CC774@EXCH125.IFN.com>
Message-ID: <SNT127-W263D08BF2606CDBE39E64FD5250@phx.gbl>


you can try to launch such lua script via bgapi. Also you can send variables you need via args.bgapi lua scriptname.lua arg1 arg2 etc...
From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
CC: nasida at live.ru
Subject: RE: [Freeswitch-users] File upload - mod_curl
Date: Fri, 27 Apr 2012 19:18:07 +0000

In my use I am using outbound socket for FAXAfter receiving a ?CUTOM - spandsp::rxfaxresult ? event , I need to call curl/http application  to upload the received fax from remote FreeSWITCH to storage.I want to run the file upload process in background ? bgapi curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log. RegardsRaviRaj. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] File upload - mod_curl For example , you can to launch some lua script from FS dialplan and invoke linux command from lua via  io.popen()http://wiki.freeswitch.org/wiki/Mod_lua#Run_a_shell_command From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curlHi  Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH. From Linux command prompt SUCCESScurl -F file=@/tmp/log/callcontrol.log http://192.168.9.152/upload  From fs_cli curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.logcurl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log All the above results in <html><head><title>415 Unsupported Media Type</title></head><body bgcolor="white"><center><h1>415 Unsupported Media Type</h1></center><hr><center>nginx/1.2.0</center></body></html> FreeSWITCH versionFreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000) ThanksRaviRaj.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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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From nasida at live.ru  Sat Apr 28 00:03:43 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sat, 28 Apr 2012 00:03:43 +0400
Subject: [Freeswitch-users] SIP invite contact header... again
In-Reply-To: <4F9913A5.505@ricozome.net>
References: <4F9913A5.505@ricozome.net>
Message-ID: <SNT127-W237EF17AF2F8A06696208ED5250@phx.gbl>


Usually caller phone number transferring in "From:" header or in "Remote-Party-ID"http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
orhttp://wiki.freeswitch.org/wiki/Variable_sip_cid_type

use it from dialplan.
If you still want to change "Contact:", try use it:http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Changing_the_SIP_Contact_user 


> Date: Thu, 26 Apr 2012 11:21:41 +0200
> From: rico-freeswitch at ricozome.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] SIP invite contact header... again
> 
> Hi list,
> 
> I have some analogic phones (to be axact two Siemens Gigaset DECT
> phones) connected to my FS server through a RTC-to-SIP gateway.
> 
> It is working well, except for announcing the caller phone number. After
> some investigations, it appear the SIP gateway expect the caller number
> into the "Contact" SIP header like this :
> 
> Contact: <sip:105002 at 46.28.168.58:5060>
> 
> I found a lot of documentation related to manipulating SIP header on
> gateways, but in that particular case, I need to alter SIP contact
> header from and to a particular extension, and most of call don't even
> transit through a SIP gateway as they are handled into the same SIP profile.
> 
> I guess this is trivial, but I'm still a newbie on IPBX and FS, and
> explanations I found on the wiki are too obsure for me...
> 
> Thanks for your help,
> 
> -- 
> Eric
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From nasida at live.ru  Sat Apr 28 00:04:38 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sat, 28 Apr 2012 00:04:38 +0400
Subject: [Freeswitch-users] SIP invite contact header... again
In-Reply-To: <4F9913A5.505@ricozome.net>
References: <4F9913A5.505@ricozome.net>
Message-ID: <SNT127-W23624D68E59AF8201B68DBD5250@phx.gbl>



Usually caller phone number transferring in "From:" header or in "Remote-Party-ID"http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
orhttp://wiki.freeswitch.org/wiki/Variable_sip_cid_type

use it from dialplan.
If you still want to change "Contact:", try use it:http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Changing_the_SIP_Contact_user 


> Date: Thu, 26 Apr 2012 11:21:41 +0200
> From: rico-freeswitch at ricozome.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] SIP invite contact header... again
> 
> Hi list,
> 
> I have some analogic phones (to be axact two Siemens Gigaset DECT
> phones) connected to my FS server through a RTC-to-SIP gateway.
> 
> It is working well, except for announcing the caller phone number. After
> some investigations, it appear the SIP gateway expect the caller number
> into the "Contact" SIP header like this :
> 
> Contact: <sip:105002 at 46.28.168.58:5060>
> 
> I found a lot of documentation related to manipulating SIP header on
> gateways, but in that particular case, I need to alter SIP contact
> header from and to a particular extension, and most of call don't even
> transit through a SIP gateway as they are handled into the same SIP profile.
> 
> I guess this is trivial, but I'm still a newbie on IPBX and FS, and
> explanations I found on the wiki are too obsure for me...
> 
> Thanks for your help,
> 
> -- 
> Eric
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From tlanghals at gmail.com  Sat Apr 28 02:01:58 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Fri, 27 Apr 2012 17:01:58 -0500
Subject: [Freeswitch-users] Forward incoming call to cell phone if user not
	reachable/registered
Message-ID: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>

Could someone please help with an example dialplan for call failover to an
external DID (i.e. cell phone or pots line) in case the the user has a
network failure can cannot be reached?
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From william.king at quentustech.com  Sat Apr 28 02:25:34 2012
From: william.king at quentustech.com (William King)
Date: Fri, 27 Apr 2012 15:25:34 -0700
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
 not reachable/registered
In-Reply-To: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
Message-ID: <4F9B1CDE.6040509@quentustech.com>

You want to do research into the continue-on-fail  and the variable
hangup after bridge.
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail

Basically you can have multiple bridge actions in an extension and the
first to successfully bridge a call will be the last action the call
will make.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 04/27/2012 03:01 PM, Travis Langhals wrote:
> Could someone please help with an example dialplan for call failover
> to an external DID (i.e. cell phone or pots line) in case the the user
> has a network failure can cannot be reached?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From rml at tollfreeforwarding.com  Sat Apr 28 03:23:24 2012
From: rml at tollfreeforwarding.com (RaviRaj Mulasa)
Date: Fri, 27 Apr 2012 23:23:24 +0000
Subject: [Freeswitch-users] File upload - mod_curl
In-Reply-To: <SNT127-W263D08BF2606CDBE39E64FD5250@phx.gbl>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>,
	<SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>,
	<8B94625BC339264DBA61E314BE9EC2CF2E9CC774@EXCH125.IFN.com>
	<SNT127-W263D08BF2606CDBE39E64FD5250@phx.gbl>
Message-ID: <8B94625BC339264DBA61E314BE9EC2CF2E9CC821@EXCH125.IFN.com>

Yes, I  understand and agree we can run a lua script for the file upload in background.
This requires mod_lua and writing a lua script for the upload.

I want to refrain from doing so as all our call control logic resides on a Java server we are developing.
We want to make clever use of mod_curl to upload a file using multipart FORM data and HTTP POST method.
There is no documentation @ http://wiki.freeswitch.org/wiki/Mod_http how to achieve this.

Regards
RaviRaj.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 12:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] File upload - mod_curl

you can try to launch such lua script via bgapi. Also you can send variables you need via args.
bgapi lua scriptname.lua arg1 arg2 etc...

________________________________
From: rml at tollfreeforwarding.com<mailto:rml at tollfreeforwarding.com>
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
CC: nasida at live.ru<mailto:nasida at live.ru>
Subject: RE: [Freeswitch-users] File upload - mod_curl
Date: Fri, 27 Apr 2012 19:18:07 +0000
In my use I am using outbound socket for FAX
After receiving a 'CUTOM - spandsp::rxfaxresult ' event , I need to call curl/http application  to upload the received fax from remote FreeSWITCH to storage.
I want to run the file upload process in background - bgapi curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>.

Regards
RaviRaj.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org]<mailto:[mailto:freeswitch-users-bounces at lists.freeswitch.org]> On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 11:57 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] File upload - mod_curl

For example , you can to launch some lua script from FS dialplan and invoke linux command from lua via  io.popen()
http://wiki.freeswitch.org/wiki/Mod_lua#Run_a_shell_command
________________________________
From: rml at tollfreeforwarding.com<mailto:rml at tollfreeforwarding.com>
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curl
Hi

Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH.

>From Linux command prompt

SUCCESS
curl -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log> http://192.168.9.152/upload


>From fs_cli

curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.log
curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>
curl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log<mailto:file=@/tmp/log/callcontrol.log>

All the above results in

<html>
<head><title>415 Unsupported Media Type</title></head>
<body bgcolor="white">
<center><h1>415 Unsupported Media Type</h1></center>
<hr><center>nginx/1.2.0</center>
</body>
</html>

FreeSWITCH version
FreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000)

Thanks
RaviRaj.

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org<mailto:consulting at freeswitch.org> http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at 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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From nasida at live.ru  Sat Apr 28 04:45:13 2012
From: nasida at live.ru (Yuriy Nasida)
Date: Sat, 28 Apr 2012 04:45:13 +0400
Subject: [Freeswitch-users] File upload - mod_curl
In-Reply-To: <8B94625BC339264DBA61E314BE9EC2CF2E9CC821@EXCH125.IFN.com>
References: <8B94625BC339264DBA61E314BE9EC2CF2E9CC5C0@EXCH125.IFN.com>, ,
	<SNT127-W8AE79B5F463F30092C8AED5250@phx.gbl>, ,
	<8B94625BC339264DBA61E314BE9EC2CF2E9CC774@EXCH125.IFN.com>,
	<SNT127-W263D08BF2606CDBE39E64FD5250@phx.gbl>,
	<8B94625BC339264DBA61E314BE9EC2CF2E9CC821@EXCH125.IFN.com>
Message-ID: <SNT127-W65D57D1239E55085BAA925D52A0@phx.gbl>


Not sure if it is possible. I had some issue with mod_curl  in the past.  I needed to set max time for curl request but  mod_curl  had not it. That is why i used linux curl via lua. I mean that  mod_curl can be insufficiently flexible for your goal. However it is only my assumption. For full understanding, you need to read source code of this module or wait for other advise.

From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
CC: nasida at live.ru
Subject: RE: [Freeswitch-users] File upload - mod_curl
Date: Fri, 27 Apr 2012 23:23:24 +0000

Yes, I  understand and agree we can run a lua script for the file upload in background.This requires mod_lua and writing a lua script for the upload. I want to refrain from doing so as all our call control logic resides on a Java server we are developing.We want to make clever use of mod_curl to upload a file using multipart FORM data and HTTP POST method.There is no documentation @ http://wiki.freeswitch.org/wiki/Mod_http how to achieve this. RegardsRaviRaj. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 12:40 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] File upload - mod_curl you can try to launch such lua script via bgapi. Also you can send variables you need via args.bgapi lua scriptname.lua arg1 arg2 etc... From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
CC: nasida at live.ru
Subject: RE: [Freeswitch-users] File upload - mod_curl
Date: Fri, 27 Apr 2012 19:18:07 +0000In my use I am using outbound socket for FAXAfter receiving a ?CUTOM - spandsp::rxfaxresult ? event , I need to call curl/http application  to upload the received fax from remote FreeSWITCH to storage.I want to run the file upload process in background ? bgapi curl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.log. RegardsRaviRaj. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida
Sent: Friday, April 27, 2012 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] File upload - mod_curl For example , you can to launch some lua script from FS dialplan and invoke linux command from lua via  io.popen()http://wiki.freeswitch.org/wiki/Mod_lua#Run_a_shell_command From: rml at tollfreeforwarding.com
To: freeswitch-users at lists.freeswitch.org
Date: Fri, 27 Apr 2012 00:01:30 +0000
Subject: [Freeswitch-users] File upload - mod_curlHi  Please let me know how can I upload a file from fs_cli using curl or other http application of FreeSWITCH. From Linux command prompt SUCCESScurl -F file=@/tmp/log/callcontrol.log http://192.168.9.152/upload  From fs_cli curl http://192.168.9.152/upload post file=%40%2Ftmp%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=%40%2Ftmp%2Fccv5%2Flog%2Fcallcontrol.logcurl http://192.168.9.152/upload post -F file=@/tmp/log/callcontrol.logcurl http://192.168.9.152/upload post file=@/tmp/log/callcontrol.log All the above results in <html><head><title>415 Unsupported Media Type</title></head><body bgcolor="white"><center><h1>415 Unsupported Media Type</h1></center><hr><center>nginx/1.2.0</center></body></html> FreeSWITCH versionFreeSWITCH Version 1.1.beta1 (git-65c4f76 2012-04-25 17-58-41 +0000) ThanksRaviRaj.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Sat Apr 28 05:05:55 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 27 Apr 2012 20:05:55 -0500
Subject: [Freeswitch-users] Eaves Drop
In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local>
References: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local>
Message-ID: <CAKbxfG9oh_m8VfFgeZOpGnWJGJ-GZ4B=iE0ww2xr4N3mEb2nAg@mail.gmail.com>

There are commands to list and clear media bugs from a certian uuid.  I
forgot the name but you should see it in mod commands.
On Apr 27, 2012 6:35 AM, "Rob Moore" <Rob.Moore at aeriandi.com> wrote:

>  Hi All,****
>
> ** **
>
> Hopefully looking for a quick answer here. We are using the Eavesdrop
> function to supply a ?call barging? service for a call centre we are
> running through a web UI but we have a few problems around transfers where
> we park the call and switching the barge from one agent to another.****
>
> ** **
>
> Basically I need some method of removing the media bug from the agents
> channel. I?ve searched the Wiki and done a few googles but I can?t seem to
> find anything that we can call to remove a bug.****
>
> ** **
>
> We?re doing all of our control of freeswitch through lua so anything that
> we can call from the CLI would be great.****
>
> ** **
>
> Anyone have any suggestions?****
>
> ** **
>
> Thanks in advance!****
>
> ** **
>
> Rob****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bdfoster at endigotech.com  Sat Apr 28 06:39:24 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Fri, 27 Apr 2012 22:39:24 -0400
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
 not reachable/registered
In-Reply-To: <4F9B1CDE.6040509@quentustech.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
	<4F9B1CDE.6040509@quentustech.com>
Message-ID: <CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>

Look at the local exensions dialplan in default.xml for an example,
specifically how it is forwarded on to voicemail. You could get slick and
do conditions and set a param inside of the users .xml file in the
directory to tell the dialplan where to try next. Also look at the
call_timeout variable.

-BDF
On Apr 27, 2012 6:26 PM, "William King" <william.king at quentustech.com>
wrote:

>  You want to do research into the continue-on-fail  and the variable
> hangup after bridge.
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> Basically you can have multiple bridge actions in an extension and the
> first to successfully bridge a call will be the last action the call will
> make.
>
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518william.king at quentustech.com
>
>
> On 04/27/2012 03:01 PM, Travis Langhals wrote:
>
> Could someone please help with an example dialplan for call failover to an
> external DID (i.e. cell phone or pots line) in case the the user has a
> network failure can cannot be reached?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From aahuta at gmail.com  Sat Apr 28 00:29:17 2012
From: aahuta at gmail.com (Alfred Ahuta)
Date: Fri, 27 Apr 2012 15:29:17 -0500
Subject: [Freeswitch-users] No Audio
Message-ID: <CACocZt8bjNCEVFfWWmnvka5M-iXxP4ZQAD3gXUOq3RwmUGU95w@mail.gmail.com>

I'm experiencing No Audio when i call pstn from an extension mapped to
X-lite. internal sip endpoints i'm able to establish 2 way audio
but not when i call out. i'm able to establish connection without the audio
though.

below is my debug log
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:4918 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3017 Set Codec
sofia/external/18178455922 PCMU/8000 20 ms 160 samples 64000 bits
2012-04-27 12:57:13.533748 [DEBUG] switch_core_codec.c:111
sofia/external/18178455922 Original read codec set to PCMU:0
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:5032 Set 2833 dtmf send
payload to 101
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
[sofia/external/18178455922] XX.XXX.XX.149 port 26586 -> 67.215.226.150
port 19474 codec: 0 ms: 20
2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
payload to 101
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
payload to 101
2012-04-27 12:57:13.533748 [NOTICE] sofia_glue.c:4063 Pre-Answer
sofia/external/18178455922!
2012-04-27 12:57:13.533748 [DEBUG] switch_channel.c:2986
(sofia/external/18178455922) Callstate Change RINGING -> EARLY
2012-04-27 12:57:13.533748 [INFO] switch_ivr_originate.c:3215 Sending early
media
2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
[sofia/internal/1000 at XX.XXX.XX.149:5080] XX.XXX.XX.149 port 22578 ->
76.187.91.204 port 40032 codec: 0 ms: 20
2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
160 bytes per 20ms
2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
payload to 101
2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
payload to 101
2012-04-27 12:57:13.553797 [DEBUG] mod_sofia.c:2582 Ring SDP:
v=0
o=FreeSWITCH 1335523255 1335523256 IN IP4 XX.XXX.XX.149
s=FreeSWITCH
c=IN IP4 XX.XXX.XX.149
t=0 0
m=audio 22578 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

o=root 2889 2889 IN IP4 67.215.226.150
s=session
c=IN IP4 67.215.226.150
t=0 0
m=audio 12116 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20






2012-04-27 12:57:13.553797 [NOTICE] mod_sofia.c:2585 Pre-Answer
sofia/internal/1000 at XX.XXX.XX.149:5080!
2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:2986
(sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change RINGING -> EARLY
2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:3028 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:13.553797 [DEBUG] sofia.c:5554 Channel
sofia/internal/1000 at XX.XXX.XX.149:5080 skipping state [early][183]
2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:13.553797 [DEBUG] switch_ivr_originate.c:3266 Originate
Resulted in Success: [sofia/external/18178455922]
2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:13.573784 [DEBUG] switch_ivr_bridge.c:1331
(sofia/external/18178455922) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:362
(sofia/external/18178455922) Running State Change CS_EXCHANGE_MEDIA
2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:1182 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:420
(sofia/external/18178455922) State EXCHANGE_MEDIA
2012-04-27 12:57:13.573784 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
sofia/external/18178455922 entering state [completing][200]
2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5572 Remote SDP:
v=0
o=root 2889 2890 IN IP4 67.215.226.150
s=session
c=IN IP4 67.215.226.150
t=0 0
m=audio 19474 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
sofia/external/18178455922 entering state [ready][200]
2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3245
(sofia/external/18178455922) Callstate Change EARLY -> ACTIVE
2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3257 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.513750 [NOTICE] sofia.c:6214 Channel
[sofia/external/18178455922] has been answered
2012-04-27 12:57:17.533751 [DEBUG] mod_sofia.c:754 Local SDP
sofia/internal/1000 at XX.XXX.XX.149:5080:
v=0
o=FreeSWITCH 1335523255 1335523257 IN IP4 XX.XXX.XX.149
s=FreeSWITCH
c=IN IP4 XX.XXX.XX.149
t=0 0
m=audio 22578 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.533751 [DEBUG] sofia.c:5561 Channel
sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [completed][200]
2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.533751 [DEBUG] switch_channel.c:3245
(sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change EARLY -> ACTIVE
2012-04-27 12:57:17.533751 [NOTICE] switch_ivr_bridge.c:417 Channel
[sofia/internal/1000 at XX.XXX.XX.149:5080] has been answered
2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:17.713754 [DEBUG] sofia.c:5561 Channel
sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [ready][200]
2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.913748 [DEBUG] switch_core_session.c:877 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2849
(sofia/external/18178455922) Callstate Change ACTIVE -> HANGUP
2012-04-27 12:57:24.933750 [NOTICE] sofia.c:636 Hangup
sofia/external/18178455922 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2872 Send signal
sofia/external/18178455922 [KILL]
2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
DONE [sofia/external/18178455922]
2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:611 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:420
(sofia/external/18178455922) State EXCHANGE_MEDIA going to sleep
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
(sofia/external/18178455922) Running State Change CS_HANGUP
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
(sofia/external/18178455922) State HANGUP
2012-04-27 12:57:24.933750 [DEBUG] mod_sofia.c:469 Channel
sofia/external/18178455922 hanging up, cause: NORMAL_CLEARING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:47
sofia/external/18178455922 Standard HANGUP, cause: NORMAL_CLEARING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
(sofia/external/18178455922) State HANGUP going to sleep
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:393
(sofia/external/18178455922) State Change CS_HANGUP -> CS_REPORTING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
(sofia/external/18178455922) Running State Change CS_REPORTING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
(sofia/external/18178455922) State REPORTING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:79
sofia/external/18178455922 Standard REPORTING, cause: NORMAL_CLEARING
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
(sofia/external/18178455922) State REPORTING going to sleep
2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:387
(sofia/external/18178455922) State Change CS_REPORTING -> CS_DESTROY
2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1382 Session 205
(sofia/external/18178455922) Locked, Waiting on external entities
2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
DONE [sofia/internal/1000 at XX.XXX.XX.149:5080]
2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:611 Send signal
sofia/external/18178455922 [BREAK]
2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:731 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:226
sofia/internal/1000 at XX.XXX.XX.149:5080 has executed the last dialplan
instruction, hanging up.
2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2849
(sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change ACTIVE -> HANGUP
2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:228 Hangup
sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_EXECUTE] [NORMAL_CLEARING]
2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2872 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [KILL]
2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:417
(sofia/internal/1000 at XX.XXX.XX.149:5080) State EXECUTE going to sleep
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_HANGUP
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP
2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:463
sofia/internal/1000 at XX.XXX.XX.149:5080 Overriding SIP cause 480 with 200
from the other leg
2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:469 Channel
sofia/internal/1000 at XX.XXX.XX.149:5080 hanging up, cause: NORMAL_CLEARING
2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:517 Sending BYE to
sofia/internal/1000 at XX.XXX.XX.149:5080
2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1400 Session 205
(sofia/external/18178455922) Ended
2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/external/18178455922 [CS_DESTROY]
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:491
(sofia/external/18178455922) Callstate Change HANGUP -> DOWN
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:494
(sofia/external/18178455922) Running State Change CS_DESTROY
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
(sofia/external/18178455922) State DESTROY
2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:374
sofia/external/18178455922 SOFIA DESTROY
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:86
sofia/external/18178455922 Standard DESTROY
2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
(sofia/external/18178455922) State DESTROY going to sleep
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:47
sofia/internal/1000 at XX.XXX.XX.149:5080 Standard HANGUP, cause:
NORMAL_CLEARING
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:602
(sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP going to sleep
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:393
(sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_HANGUP ->
CS_REPORTING
2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:362
(sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_REPORTING
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:79
sofia/internal/1000 at XX.XXX.XX.149:5080 Standard REPORTING, cause:
NORMAL_CLEARING
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
(sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING going to sleep
2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:387
(sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_REPORTING ->
CS_DESTROY
2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1382 Session 204
(sofia/internal/1000 at XX.XXX.XX.149:5080) Locked, Waiting on external
entities
2012-04-27 12:57:24.973781 [NOTICE] switch_core_session.c:1400 Session 204
(sofia/internal/1000 at XX.XXX.XX.149:5080) Ended
2012-04-27 12:57:24.995020 [NOTICE] switch_core_session.c:1402 Close
Channel sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_DESTROY]
2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change HANGUP -> DOWN
2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:494
(sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_DESTROY
2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY
2012-04-27 12:57:24.995020 [DEBUG] mod_sofia.c:374
sofia/internal/1000 at XX.XXX.XX.149:5080 SOFIA DESTROY
2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:86
sofia/internal/1000 at XX.XXX.XX.149:5080 Standard DESTROY
2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
(sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY going to sleep


Alfred
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From anton.jugatsu at gmail.com  Sat Apr 28 12:02:39 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 28 Apr 2012 12:02:39 +0400
Subject: [Freeswitch-users] No Audio
In-Reply-To: <CACocZt8bjNCEVFfWWmnvka5M-iXxP4ZQAD3gXUOq3RwmUGU95w@mail.gmail.com>
References: <CACocZt8bjNCEVFfWWmnvka5M-iXxP4ZQAD3gXUOq3RwmUGU95w@mail.gmail.com>
Message-ID: <CANBhyPD4L85WQ17XOzT4fgdtvwaz8hYZLMmfpski599_YZF5UA@mail.gmail.com>

What do use for PSTN termination? VOIP-gateway or just sip trunk? Show your
dialplan for outgoing to PSTN.

28 ?????? 2012 ?. 0:29 ???????????? Alfred Ahuta <aahuta at gmail.com> ???????:

> I'm experiencing No Audio when i call pstn from an extension mapped to
> X-lite. internal sip endpoints i'm able to establish 2 way audio
> but not when i call out. i'm able to establish connection without the
> audio though.
>
> below is my debug log
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:4918 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3017 Set Codec
> sofia/external/18178455922 PCMU/8000 20 ms 160 samples 64000 bits
> 2012-04-27 12:57:13.533748 [DEBUG] switch_core_codec.c:111
> sofia/external/18178455922 Original read codec set to PCMU:0
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:5032 Set 2833 dtmf send
> payload to 101
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
> [sofia/external/18178455922] XX.XXX.XX.149 port 26586 -> 67.215.226.150
> port 19474 codec: 0 ms: 20
> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
> payload to 101
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
> payload to 101
> 2012-04-27 12:57:13.533748 [NOTICE] sofia_glue.c:4063 Pre-Answer
> sofia/external/18178455922!
> 2012-04-27 12:57:13.533748 [DEBUG] switch_channel.c:2986
> (sofia/external/18178455922) Callstate Change RINGING -> EARLY
> 2012-04-27 12:57:13.533748 [INFO] switch_ivr_originate.c:3215 Sending
> early media
> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
> [sofia/internal/1000 at XX.XXX.XX.149:5080] XX.XXX.XX.149 port 22578 ->
> 76.187.91.204 port 40032 codec: 0 ms: 20
> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer [soft]
> 160 bytes per 20ms
> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
> payload to 101
> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive
> payload to 101
> 2012-04-27 12:57:13.553797 [DEBUG] mod_sofia.c:2582 Ring SDP:
> v=0
> o=FreeSWITCH 1335523255 1335523256 IN IP4 XX.XXX.XX.149
> s=FreeSWITCH
> c=IN IP4 XX.XXX.XX.149
> t=0 0
> m=audio 22578 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> o=root 2889 2889 IN IP4 67.215.226.150
> s=session
> c=IN IP4 67.215.226.150
> t=0 0
> m=audio 12116 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
>
>
>
>
>
> 2012-04-27 12:57:13.553797 [NOTICE] mod_sofia.c:2585 Pre-Answer
> sofia/internal/1000 at XX.XXX.XX.149:5080!
> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:2986
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change RINGING -> EARLY
> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:3028 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:13.553797 [DEBUG] sofia.c:5554 Channel
> sofia/internal/1000 at XX.XXX.XX.149:5080 skipping state [early][183]
> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:13.553797 [DEBUG] switch_ivr_originate.c:3266 Originate
> Resulted in Success: [sofia/external/18178455922]
> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:13.573784 [DEBUG] switch_ivr_bridge.c:1331
> (sofia/external/18178455922) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/18178455922) Running State Change CS_EXCHANGE_MEDIA
> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:420
> (sofia/external/18178455922) State EXCHANGE_MEDIA
> 2012-04-27 12:57:13.573784 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
> sofia/external/18178455922 entering state [completing][200]
> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5572 Remote SDP:
> v=0
> o=root 2889 2890 IN IP4 67.215.226.150
> s=session
> c=IN IP4 67.215.226.150
> t=0 0
> m=audio 19474 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
> sofia/external/18178455922 entering state [ready][200]
> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3245
> (sofia/external/18178455922) Callstate Change EARLY -> ACTIVE
> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3257 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.513750 [NOTICE] sofia.c:6214 Channel
> [sofia/external/18178455922] has been answered
> 2012-04-27 12:57:17.533751 [DEBUG] mod_sofia.c:754 Local SDP
> sofia/internal/1000 at XX.XXX.XX.149:5080:
> v=0
> o=FreeSWITCH 1335523255 1335523257 IN IP4 XX.XXX.XX.149
> s=FreeSWITCH
> c=IN IP4 XX.XXX.XX.149
> t=0 0
> m=audio 22578 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.533751 [DEBUG] sofia.c:5561 Channel
> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [completed][200]
> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.533751 [DEBUG] switch_channel.c:3245
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change EARLY -> ACTIVE
> 2012-04-27 12:57:17.533751 [NOTICE] switch_ivr_bridge.c:417 Channel
> [sofia/internal/1000 at XX.XXX.XX.149:5080] has been answered
> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:17.713754 [DEBUG] sofia.c:5561 Channel
> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [ready][200]
> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.913748 [DEBUG] switch_core_session.c:877 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2849
> (sofia/external/18178455922) Callstate Change ACTIVE -> HANGUP
> 2012-04-27 12:57:24.933750 [NOTICE] sofia.c:636 Hangup
> sofia/external/18178455922 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2872 Send signal
> sofia/external/18178455922 [KILL]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
> DONE [sofia/external/18178455922]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:611 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:420
> (sofia/external/18178455922) State EXCHANGE_MEDIA going to sleep
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/18178455922) Running State Change CS_HANGUP
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
> (sofia/external/18178455922) State HANGUP
> 2012-04-27 12:57:24.933750 [DEBUG] mod_sofia.c:469 Channel
> sofia/external/18178455922 hanging up, cause: NORMAL_CLEARING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:47
> sofia/external/18178455922 Standard HANGUP, cause: NORMAL_CLEARING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
> (sofia/external/18178455922) State HANGUP going to sleep
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:393
> (sofia/external/18178455922) State Change CS_HANGUP -> CS_REPORTING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
> (sofia/external/18178455922) Running State Change CS_REPORTING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
> (sofia/external/18178455922) State REPORTING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:79
> sofia/external/18178455922 Standard REPORTING, cause: NORMAL_CLEARING
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
> (sofia/external/18178455922) State REPORTING going to sleep
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:387
> (sofia/external/18178455922) State Change CS_REPORTING -> CS_DESTROY
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1382 Session 205
> (sofia/external/18178455922) Locked, Waiting on external entities
> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
> DONE [sofia/internal/1000 at XX.XXX.XX.149:5080]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:611 Send signal
> sofia/external/18178455922 [BREAK]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:731 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:226
> sofia/internal/1000 at XX.XXX.XX.149:5080 has executed the last dialplan
> instruction, hanging up.
> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2849
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change ACTIVE -> HANGUP
> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:228 Hangup
> sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_EXECUTE] [NORMAL_CLEARING]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2872 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [KILL]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:417
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State EXECUTE going to sleep
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_HANGUP
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP
> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:463
> sofia/internal/1000 at XX.XXX.XX.149:5080 Overriding SIP cause 480 with 200
> from the other leg
> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:469 Channel
> sofia/internal/1000 at XX.XXX.XX.149:5080 hanging up, cause: NORMAL_CLEARING
> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:517 Sending BYE to
> sofia/internal/1000 at XX.XXX.XX.149:5080
> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1400 Session 205
> (sofia/external/18178455922) Ended
> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1402 Close
> Channel sofia/external/18178455922 [CS_DESTROY]
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:491
> (sofia/external/18178455922) Callstate Change HANGUP -> DOWN
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:494
> (sofia/external/18178455922) Running State Change CS_DESTROY
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
> (sofia/external/18178455922) State DESTROY
> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:374
> sofia/external/18178455922 SOFIA DESTROY
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:86
> sofia/external/18178455922 Standard DESTROY
> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
> (sofia/external/18178455922) State DESTROY going to sleep
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:47
> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:602
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP going to sleep
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:393
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_HANGUP ->
> CS_REPORTING
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_REPORTING
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:79
> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING going to sleep
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:387
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_REPORTING ->
> CS_DESTROY
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1382 Session 204
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Locked, Waiting on external
> entities
> 2012-04-27 12:57:24.973781 [NOTICE] switch_core_session.c:1400 Session 204
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Ended
> 2012-04-27 12:57:24.995020 [NOTICE] switch_core_session.c:1402 Close
> Channel sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_DESTROY]
> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change HANGUP -> DOWN
> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:494
> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_DESTROY
> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY
> 2012-04-27 12:57:24.995020 [DEBUG] mod_sofia.c:374
> sofia/internal/1000 at XX.XXX.XX.149:5080 SOFIA DESTROY
> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:86
> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard DESTROY
> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY going to sleep
>
>
> Alfred
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From bernard.david.murphy at gmail.com  Sat Apr 28 13:53:47 2012
From: bernard.david.murphy at gmail.com (Bernard Murphy)
Date: Sat, 28 Apr 2012 10:53:47 +0100
Subject: [Freeswitch-users] rfc2833 first Marker Bit not set to True on bleg
Message-ID: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>

On the very latest GIT and previous , Freeswitch doesn't seem to set the
first Marker Bit to true on an rfc2833 event on the bleg of a call out to a
gateway

on the A-leg the first event from my UA to Freeswitch is correct with
duration as 0 and Marker bit set to True. Then the second event increments
the duration and Market bit is set to false.
but on the B-leg from freeswitch to the cisco gateway the first event has
duration 80 (may or may not be a problem) but the Marker bit is set to
False which must be wrong ? Duration then increments and Marker bit remains
false.

I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/> to
make the duration start at zero but it seemed to have no effect.


*ALEG from SIP UA to FREESWITCH*
No.     Time Source                Destination           Protocol Length
Info
   5254 23   10.0.47.110           10.50.47.30           RTP EVENT 60
Payload type=RTP Event, DTMF One 1

Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
Vmware_ab:60:ec (00:0c:29:ab:60:ec)
Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
10.50.47.30 (10.50.47.30)
User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
Real-Time Transport Protocol
    [Stream setup by SDP (frame 1122)]
        [Setup frame: 1122]
        [Setup Method: SDP]
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 .... = Extension: False
    .... 0000 = Contributing source identifiers count: 0
    1... .... = Marker: True
    Payload type: telephone-event (101)
    Sequence number: 26364
    [Extended sequence number: 91900]
    Timestamp: 133920
    Synchronization Source identifier: 0x03bf0676 (62850678)
RFC 2833 RTP Event
    Event ID: DTMF One 1 (1)
    0... .... = End of Event: False
    .0.. .... = Reserved: False
    ..00 1010 = Volume: 10
    Event Duration: 0

*ALEG SECOND EVENT FROM SIP UA TO FREESWITCH*

No.     Time Source                Destination           Protocol Length
Info
   5255 23   10.0.47.110           10.50.47.30           RTP EVENT 60
Payload type=RTP Event, DTMF One 1

Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
Vmware_ab:60:ec (00:0c:29:ab:60:ec)
Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
10.50.47.30 (10.50.47.30)
User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
Real-Time Transport Protocol
    [Stream setup by SDP (frame 1122)]
        [Setup frame: 1122]
        [Setup Method: SDP]
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 .... = Extension: False
    .... 0000 = Contributing source identifiers count: 0
    0... .... = Marker: False
    Payload type: telephone-event (101)
    Sequence number: 26365
    [Extended sequence number: 91901]
    Timestamp: 133920
    Synchronization Source identifier: 0x03bf0676 (62850678)
RFC 2833 RTP Event
    Event ID: DTMF One 1 (1)
    0... .... = End of Event: False
    .0.. .... = Reserved: False
    ..00 1010 = Volume: 10
    Event Duration: 160

*BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO*
No.     Time Source                Destination           Protocol Length
Info
   5329 23   10.50.47.30           10.50.47.19           RTP EVENT 58
Payload type=RTP Event, DTMF One 1

Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
10.50.47.19 (10.50.47.19)
User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
Real-Time Transport Protocol
    [Stream setup by SDP (frame 1113)]
        [Setup frame: 1113]
        [Setup Method: SDP]
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 .... = Extension: False
    .... 0000 = Contributing source identifiers count: 0
    0... .... = Marker: False
    Payload type: telephone-event (101)
    Sequence number: 24716
    [Extended sequence number: 90252]
    Timestamp: 158240
    Synchronization Source identifier: 0x57e5d7a1 (1474680737)
RFC 2833 RTP Event
    Event ID: DTMF One 1 (1)
    0... .... = End of Event: False
    .0.. .... = Reserved: False
    ..00 0111 = Volume: 7
    Event Duration: 80

*BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO*
No.     Time Source                Destination           Protocol Length
Info
   5331 23   10.50.47.30           10.50.47.19           RTP EVENT 58
Payload type=RTP Event, DTMF One 1

Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
10.50.47.19 (10.50.47.19)
User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
Real-Time Transport Protocol
    [Stream setup by SDP (frame 1113)]
        [Setup frame: 1113]
        [Setup Method: SDP]
    10.. .... = Version: RFC 1889 Version (2)
    ..0. .... = Padding: False
    ...0 .... = Extension: False
    .... 0000 = Contributing source identifiers count: 0
    0... .... = Marker: False
    Payload type: telephone-event (101)
    Sequence number: 24717
    [Extended sequence number: 90253]
    Timestamp: 158240
    Synchronization Source identifier: 0x57e5d7a1 (1474680737)
RFC 2833 RTP Event
    Event ID: DTMF One 1 (1)
    0... .... = End of Event: False
    .0.. .... = Reserved: False
    ..00 0111 = Volume: 7
    Event Duration: 160
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From mario at ims.bg  Sat Apr 28 14:08:20 2012
From: mario at ims.bg (Mario Karakanovski)
Date: Sat, 28 Apr 2012 13:08:20 +0300
Subject: [Freeswitch-users] client change IP address while ringing
In-Reply-To: <4F9AC5D5.3030509@communicatefreely.net>
References: <B4FE049A6F1C45FC859AFCF6D3DFD080@imsdeveloper>	<CALn233qM80E8izqKwkyM9sqEg9sJ_2Np-HoJ8FHbZN6683TYKQ@mail.gmail.com><BCCA7469823E440096A6D6BC76E07037@imsdeveloper>
	<4F9AC5D5.3030509@communicatefreely.net>
Message-ID: <550F9162C4084345BF440BA334A893D2@imsdeveloper>

Thank you Tim,

	It was really helpful, but it is not impossible I think. 

I solved the problem for changing network in a call. What's I am doing is
just send RE-INVAITE from the client. Yes, there is a small gap (< 1sec.),
but it is acceptable. When this gap is used to inform subscriber for
changing network the things are going pretty well. Of course, the client
sends REGISTER as far as possible.

So, from my experience: sending RE-INVITE with SDP is enough to force
freeswitch to send media to correct destination. Changing the registration
in a call will not force freeswitch to send BYE to correct destination after
bridge ends (maybe it is a bug, maybe not, I am not sure). 

The problem that I should to solve is when client changes his network
between RING and ANSWER.
So, freeswitch will receive new registration, but how to close existing
bridge in case when client change his network?
In this case RE-INVITE does not work.

The idea I have, but not tested is to send BYE from the client, with the
same Call-ID and some special reason. After that maybe I will be able to
bridge call again with correct parameters.

Regards,
	Mario

 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim St.
Pierre
Sent: Friday, April 27, 2012 7:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] client change IP address while ringing

That's an impossible situation -

How does FS know the new address?  The only way is if the phone 
registers.  FS won't even know where to send the next packet until the 
phone does another registration so it gets an update.

If you have a registration for both the WiFi and 3G address, you will 
have better luck, but I think FS will still maintain the contact info 
for a dialog that has already started.

If this is a problem you really need to solve, then you may need to set 
up some sort of VPN so that a common IP address can be reachable over 
both 3G and WiFi.  Even then, there is still going to be a period of 
instability when you transition, as the routing has to re-converge.

You also can't transition an active call between WiFi and 3G for the 
same reasons.  The only way to do that would be to park the call, 
re-register on the new network, then pick it up again.

Hope that's helpful.

-Tim


Mario Karakanovski wrote:
> Thank you Mitch,
>
> 	The client use TCP, but the problem is that it must change IP
> address, because it's switch between WiFi and 3G. So, there is no chance
to
> force it using the same IP address.
>
> Regards,
> 	Mario
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitch
> Capper
> Sent: Tuesday, April 24, 2012 4:24 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] client change IP address while ringing
>
> Try changing the mobile client to register via TCP I would assume then
> the IP will not change as that would break the TCP connection:)
>
> ~Mitch
>
> Hi guys,
>
>  
>
>             I have an interesting situation.
>
> I have two mobile clients and connect them using bridge. It is working
> without problems, except when the called party changes its IP address
> between ringing and answer.
>
> Does anybody deal with such situation?
>
>  
>
> Regards,
>
>             Mario  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From aahuta at gmail.com  Sat Apr 28 15:14:11 2012
From: aahuta at gmail.com (Alfred Ahuta)
Date: Sat, 28 Apr 2012 06:14:11 -0500
Subject: [Freeswitch-users] No Audio
In-Reply-To: <CANBhyPD4L85WQ17XOzT4fgdtvwaz8hYZLMmfpski599_YZF5UA@mail.gmail.com>
References: <CACocZt8bjNCEVFfWWmnvka5M-iXxP4ZQAD3gXUOq3RwmUGU95w@mail.gmail.com>
	<CANBhyPD4L85WQ17XOzT4fgdtvwaz8hYZLMmfpski599_YZF5UA@mail.gmail.com>
Message-ID: <CACocZt9UvYJtYKObVUnxDLUwyRKFQcsuw99yDutmKTRpUKGSSw@mail.gmail.com>

I'm using a sip trunk
Below is my dialplan

<include>
  <extension name="Flowroute">
    <condition field="destination_number" expression="^99(\d+)$">
          <action application="bridge" data="sofia/gateway/Flowroute/$1"/>
    </condition>
  </extension>
</include>

Thanks



2012/4/28 Anton Kvashenkin <anton.jugatsu at gmail.com>

> What do use for PSTN termination? VOIP-gateway or just sip trunk? Show
> your dialplan for outgoing to PSTN.
>
> 28 ?????? 2012 ?. 0:29 ???????????? Alfred Ahuta <aahuta at gmail.com>???????:
>
>> I'm experiencing No Audio when i call pstn from an extension mapped to
>> X-lite. internal sip endpoints i'm able to establish 2 way audio
>> but not when i call out. i'm able to establish connection without the
>> audio though.
>>
>> below is my debug log
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:4918 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3017 Set Codec
>> sofia/external/18178455922 PCMU/8000 20 ms 160 samples 64000 bits
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_core_codec.c:111 sofia/external/
>> 18178455922 Original read codec set to PCMU:0
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:5032 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
>> [sofia/external/18178455922] XX.XXX.XX.149 port 26586 -> 67.215.226.150
>> port 19474 codec: 0 ms: 20
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf
>> receive payload to 101
>> 2012-04-27 12:57:13.533748 [NOTICE] sofia_glue.c:4063 Pre-Answer
>> sofia/external/18178455922!
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_channel.c:2986 (sofia/external/
>> 18178455922) Callstate Change RINGING -> EARLY
>> 2012-04-27 12:57:13.533748 [INFO] switch_ivr_originate.c:3215 Sending
>> early media
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
>> [sofia/internal/1000 at XX.XXX.XX.149:5080] XX.XXX.XX.149 port 22578 ->
>> 76.187.91.204 port 40032 codec: 0 ms: 20
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf
>> receive payload to 101
>> 2012-04-27 12:57:13.553797 [DEBUG] mod_sofia.c:2582 Ring SDP:
>> v=0
>> o=FreeSWITCH 1335523255 1335523256 IN IP4 XX.XXX.XX.149
>> s=FreeSWITCH
>> c=IN IP4 XX.XXX.XX.149
>> t=0 0
>> m=audio 22578 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> o=root 2889 2889 IN IP4 67.215.226.150
>> s=session
>> c=IN IP4 67.215.226.150
>> t=0 0
>> m=audio 12116 RTP/AVP 0 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>>
>>
>>
>>
>> 2012-04-27 12:57:13.553797 [NOTICE] mod_sofia.c:2585 Pre-Answer
>> sofia/internal/1000 at XX.XXX.XX.149:5080!
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:2986
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change RINGING ->
>> EARLY
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:3028 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia.c:5554 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 skipping state [early][183]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_ivr_originate.c:3266 Originate
>> Resulted in Success: [sofia/external/18178455922]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_ivr_bridge.c:1331
>> (sofia/external/18178455922) State Change CS_CONSUME_MEDIA ->
>> CS_EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:420
>> (sofia/external/18178455922) State EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
>> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel sofia/external/
>> 18178455922 entering state [completing][200]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5572 Remote SDP:
>> v=0
>> o=root 2889 2890 IN IP4 67.215.226.150
>> s=session
>> c=IN IP4 67.215.226.150
>> t=0 0
>> m=audio 19474 RTP/AVP 0 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel sofia/external/
>> 18178455922 entering state [ready][200]
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3245 (sofia/external/
>> 18178455922) Callstate Change EARLY -> ACTIVE
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3257 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.513750 [NOTICE] sofia.c:6214 Channel [sofia/external/
>> 18178455922] has been answered
>> 2012-04-27 12:57:17.533751 [DEBUG] mod_sofia.c:754 Local SDP
>> sofia/internal/1000 at XX.XXX.XX.149:5080:
>> v=0
>> o=FreeSWITCH 1335523255 1335523257 IN IP4 XX.XXX.XX.149
>> s=FreeSWITCH
>> c=IN IP4 XX.XXX.XX.149
>> t=0 0
>> m=audio 22578 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.533751 [DEBUG] sofia.c:5561 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [completed][200]
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_channel.c:3245
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change EARLY -> ACTIVE
>> 2012-04-27 12:57:17.533751 [NOTICE] switch_ivr_bridge.c:417 Channel
>> [sofia/internal/1000 at XX.XXX.XX.149:5080] has been answered
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.713754 [DEBUG] sofia.c:5561 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [ready][200]
>> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.913748 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2849 (sofia/external/
>> 18178455922) Callstate Change ACTIVE -> HANGUP
>> 2012-04-27 12:57:24.933750 [NOTICE] sofia.c:636 Hangup sofia/external/
>> 18178455922 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2872 Send signal
>> sofia/external/18178455922 [KILL]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
>> DONE [sofia/external/18178455922]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:611 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:420
>> (sofia/external/18178455922) State EXCHANGE_MEDIA going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_HANGUP
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
>> (sofia/external/18178455922) State HANGUP
>> 2012-04-27 12:57:24.933750 [DEBUG] mod_sofia.c:469 Channel sofia/external/
>> 18178455922 hanging up, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:47
>> sofia/external/18178455922 Standard HANGUP, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
>> (sofia/external/18178455922) State HANGUP going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:393
>> (sofia/external/18178455922) State Change CS_HANGUP -> CS_REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
>> (sofia/external/18178455922) State REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:79
>> sofia/external/18178455922 Standard REPORTING, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
>> (sofia/external/18178455922) State REPORTING going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:387
>> (sofia/external/18178455922) State Change CS_REPORTING -> CS_DESTROY
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1382 Session 205
>> (sofia/external/18178455922) Locked, Waiting on external entities
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
>> DONE [sofia/internal/1000 at XX.XXX.XX.149:5080]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:611 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:226
>> sofia/internal/1000 at XX.XXX.XX.149:5080 has executed the last dialplan
>> instruction, hanging up.
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2849
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change ACTIVE ->
>> HANGUP
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:228
>> Hangup sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_EXECUTE]
>> [NORMAL_CLEARING]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2872 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [KILL]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:417
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State EXECUTE going to sleep
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_HANGUP
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:463
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Overriding SIP cause 480 with 200
>> from the other leg
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:469 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 hanging up, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:517 Sending BYE to
>> sofia/internal/1000 at XX.XXX.XX.149:5080
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1400 Session
>> 205 (sofia/external/18178455922) Ended
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1402 Close
>> Channel sofia/external/18178455922 [CS_DESTROY]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:491
>> (sofia/external/18178455922) Callstate Change HANGUP -> DOWN
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:494
>> (sofia/external/18178455922) Running State Change CS_DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
>> (sofia/external/18178455922) State DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:374 sofia/external/
>> 18178455922 SOFIA DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:86
>> sofia/external/18178455922 Standard DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
>> (sofia/external/18178455922) State DESTROY going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:47
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard HANGUP, cause:
>> NORMAL_CLEARING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:393
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change
>> CS_REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:79
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:387
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1382 Session 204
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Locked, Waiting on external
>> entities
>> 2012-04-27 12:57:24.973781 [NOTICE] switch_core_session.c:1400 Session
>> 204 (sofia/internal/1000 at XX.XXX.XX.149:5080) Ended
>> 2012-04-27 12:57:24.995020 [NOTICE] switch_core_session.c:1402 Close
>> Channel sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_DESTROY]
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:491
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change HANGUP -> DOWN
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:494
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] mod_sofia.c:374
>> sofia/internal/1000 at XX.XXX.XX.149:5080 SOFIA DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:86
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY going to sleep
>>
>>
>> Alfred
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From tlanghals at gmail.com  Sat Apr 28 17:46:39 2012
From: tlanghals at gmail.com (Travis Langhals)
Date: Sat, 28 Apr 2012 08:46:39 -0500
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
 not reachable/registered
In-Reply-To: <CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
	<4F9B1CDE.6040509@quentustech.com>
	<CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>
Message-ID: <CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>

Thanks for the info, I have the voicemail action working but if I add a
second bridge command between my extension bridge and voicemail actions it
rings the cell phone if the extension doesn't answer within my 20s timeout.
 I'm trying to make it only ring the cell phone if the extension is not
online and then go to voicemail.  If the extension is online it should just
ring the extension and then go to voicemail.

On Fri, Apr 27, 2012 at 9:39 PM, Brian Foster <bdfoster at endigotech.com>wrote:

> Look at the local exensions dialplan in default.xml for an example,
> specifically how it is forwarded on to voicemail. You could get slick and
> do conditions and set a param inside of the users .xml file in the
> directory to tell the dialplan where to try next. Also look at the
> call_timeout variable.
>
> -BDF
> On Apr 27, 2012 6:26 PM, "William King" <william.king at quentustech.com>
> wrote:
>
>>  You want to do research into the continue-on-fail  and the variable
>> hangup after bridge.
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>> Basically you can have multiple bridge actions in an extension and the
>> first to successfully bridge a call will be the last action the call will
>> make.
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518william.king at quentustech.com
>>
>>
>> On 04/27/2012 03:01 PM, Travis Langhals wrote:
>>
>> Could someone please help with an example dialplan for call failover to
>> an external DID (i.e. cell phone or pots line) in case the the user has a
>> network failure can cannot be reached?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anton.jugatsu at gmail.com  Sat Apr 28 21:50:04 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 28 Apr 2012 21:50:04 +0400
Subject: [Freeswitch-users] No Audio
In-Reply-To: <CANBhyPD4L85WQ17XOzT4fgdtvwaz8hYZLMmfpski599_YZF5UA@mail.gmail.com>
References: <CACocZt8bjNCEVFfWWmnvka5M-iXxP4ZQAD3gXUOq3RwmUGU95w@mail.gmail.com>
	<CANBhyPD4L85WQ17XOzT4fgdtvwaz8hYZLMmfpski599_YZF5UA@mail.gmail.com>
Message-ID: <CANBhyPA2fjRe4GVc6QK6EkHv78+yCcNp=1DJ3NkROYWdrW+3tQ@mail.gmail.com>

Is your box behind Nat? Try to play with ext-rtp-ip and ext-sip-ip for
external profile.
28.04.2012 12:02 ???????????? "Anton Kvashenkin" <anton.jugatsu at gmail.com>
???????:

> What do use for PSTN termination? VOIP-gateway or just sip trunk? Show
> your dialplan for outgoing to PSTN.
>
> 28 ?????? 2012 ?. 0:29 ???????????? Alfred Ahuta <aahuta at gmail.com>???????:
>
>> I'm experiencing No Audio when i call pstn from an extension mapped to
>> X-lite. internal sip endpoints i'm able to establish 2 way audio
>> but not when i call out. i'm able to establish connection without the
>> audio though.
>>
>> below is my debug log
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:4918 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3017 Set Codec
>> sofia/external/18178455922 PCMU/8000 20 ms 160 samples 64000 bits
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_core_codec.c:111
>> sofia/external/18178455922 Original read codec set to PCMU:0
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:5032 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
>> [sofia/external/18178455922] XX.XXX.XX.149 port 26586 -> 67.215.226.150
>> port 19474 codec: 0 ms: 20
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf
>> receive payload to 101
>> 2012-04-27 12:57:13.533748 [NOTICE] sofia_glue.c:4063 Pre-Answer
>> sofia/external/18178455922!
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_channel.c:2986
>> (sofia/external/18178455922) Callstate Change RINGING -> EARLY
>> 2012-04-27 12:57:13.533748 [INFO] switch_ivr_originate.c:3215 Sending
>> early media
>> 2012-04-27 12:57:13.533748 [DEBUG] sofia_glue.c:3269 AUDIO RTP
>> [sofia/internal/1000 at XX.XXX.XX.149:5080] XX.XXX.XX.149 port 22578 ->
>> 76.187.91.204 port 40032 codec: 0 ms: 20
>> 2012-04-27 12:57:13.533748 [DEBUG] switch_rtp.c:1661 Starting timer
>> [soft] 160 bytes per 20ms
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send
>> payload to 101
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf
>> receive payload to 101
>> 2012-04-27 12:57:13.553797 [DEBUG] mod_sofia.c:2582 Ring SDP:
>> v=0
>> o=FreeSWITCH 1335523255 1335523256 IN IP4 XX.XXX.XX.149
>> s=FreeSWITCH
>> c=IN IP4 XX.XXX.XX.149
>> t=0 0
>> m=audio 22578 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> o=root 2889 2889 IN IP4 67.215.226.150
>> s=session
>> c=IN IP4 67.215.226.150
>> t=0 0
>> m=audio 12116 RTP/AVP 0 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>>
>>
>>
>>
>>
>> 2012-04-27 12:57:13.553797 [NOTICE] mod_sofia.c:2585 Pre-Answer
>> sofia/internal/1000 at XX.XXX.XX.149:5080!
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:2986
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change RINGING ->
>> EARLY
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_channel.c:3028 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] sofia.c:5554 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 skipping state [early][183]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.553797 [DEBUG] switch_ivr_originate.c:3266 Originate
>> Resulted in Success: [sofia/external/18178455922]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_ivr_bridge.c:1331
>> (sofia/external/18178455922) State Change CS_CONSUME_MEDIA ->
>> CS_EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:13.573784 [DEBUG] switch_core_state_machine.c:420
>> (sofia/external/18178455922) State EXCHANGE_MEDIA
>> 2012-04-27 12:57:13.573784 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA
>> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.493751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
>> sofia/external/18178455922 entering state [completing][200]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5572 Remote SDP:
>> v=0
>> o=root 2889 2890 IN IP4 67.215.226.150
>> s=session
>> c=IN IP4 67.215.226.150
>> t=0 0
>> m=audio 19474 RTP/AVP 0 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.513750 [DEBUG] sofia.c:5561 Channel
>> sofia/external/18178455922 entering state [ready][200]
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3245
>> (sofia/external/18178455922) Callstate Change EARLY -> ACTIVE
>> 2012-04-27 12:57:17.513750 [DEBUG] switch_channel.c:3257 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.513750 [NOTICE] sofia.c:6214 Channel
>> [sofia/external/18178455922] has been answered
>> 2012-04-27 12:57:17.533751 [DEBUG] mod_sofia.c:754 Local SDP
>> sofia/internal/1000 at XX.XXX.XX.149:5080:
>> v=0
>> o=FreeSWITCH 1335523255 1335523257 IN IP4 XX.XXX.XX.149
>> s=FreeSWITCH
>> c=IN IP4 XX.XXX.XX.149
>> t=0 0
>> m=audio 22578 RTP/AVP 0 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.533751 [DEBUG] sofia.c:5561 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [completed][200]
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.533751 [DEBUG] switch_channel.c:3245
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change EARLY -> ACTIVE
>> 2012-04-27 12:57:17.533751 [NOTICE] switch_ivr_bridge.c:417 Channel
>> [sofia/internal/1000 at XX.XXX.XX.149:5080] has been answered
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.693753 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:17.713754 [DEBUG] sofia.c:5561 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 entering state [ready][200]
>> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:17.713754 [DEBUG] switch_core_session.c:793 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.913748 [DEBUG] switch_core_session.c:877 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2849
>> (sofia/external/18178455922) Callstate Change ACTIVE -> HANGUP
>> 2012-04-27 12:57:24.933750 [NOTICE] sofia.c:636 Hangup
>> sofia/external/18178455922 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_channel.c:2872 Send signal
>> sofia/external/18178455922 [KILL]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
>> DONE [sofia/external/18178455922]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_ivr_bridge.c:611 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:420
>> (sofia/external/18178455922) State EXCHANGE_MEDIA going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_HANGUP
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
>> (sofia/external/18178455922) State HANGUP
>> 2012-04-27 12:57:24.933750 [DEBUG] mod_sofia.c:469 Channel
>> sofia/external/18178455922 hanging up, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:47
>> sofia/external/18178455922 Standard HANGUP, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:602
>> (sofia/external/18178455922) State HANGUP going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:393
>> (sofia/external/18178455922) State Change CS_HANGUP -> CS_REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:362
>> (sofia/external/18178455922) Running State Change CS_REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
>> (sofia/external/18178455922) State REPORTING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:79
>> sofia/external/18178455922 Standard REPORTING, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:662
>> (sofia/external/18178455922) State REPORTING going to sleep
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_state_machine.c:387
>> (sofia/external/18178455922) State Change CS_REPORTING -> CS_DESTROY
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.933750 [DEBUG] switch_core_session.c:1382 Session 205
>> (sofia/external/18178455922) Locked, Waiting on external entities
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD
>> DONE [sofia/internal/1000 at XX.XXX.XX.149:5080]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_ivr_bridge.c:611 Send signal
>> sofia/external/18178455922 [BREAK]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:731 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:226
>> sofia/internal/1000 at XX.XXX.XX.149:5080 has executed the last dialplan
>> instruction, hanging up.
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2849
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change ACTIVE ->
>> HANGUP
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_state_machine.c:228
>> Hangup sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_EXECUTE]
>> [NORMAL_CLEARING]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_channel.c:2872 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [KILL]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:417
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State EXECUTE going to sleep
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_HANGUP
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:463
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Overriding SIP cause 480 with 200
>> from the other leg
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:469 Channel
>> sofia/internal/1000 at XX.XXX.XX.149:5080 hanging up, cause: NORMAL_CLEARING
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:517 Sending BYE to
>> sofia/internal/1000 at XX.XXX.XX.149:5080
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1400 Session
>> 205 (sofia/external/18178455922) Ended
>> 2012-04-27 12:57:24.954657 [NOTICE] switch_core_session.c:1402 Close
>> Channel sofia/external/18178455922 [CS_DESTROY]
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:491
>> (sofia/external/18178455922) Callstate Change HANGUP -> DOWN
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:494
>> (sofia/external/18178455922) Running State Change CS_DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
>> (sofia/external/18178455922) State DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] mod_sofia.c:374
>> sofia/external/18178455922 SOFIA DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:86
>> sofia/external/18178455922 Standard DESTROY
>> 2012-04-27 12:57:24.954657 [DEBUG] switch_core_state_machine.c:504
>> (sofia/external/18178455922) State DESTROY going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:47
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard HANGUP, cause:
>> NORMAL_CLEARING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:602
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State HANGUP going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:393
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:362
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change
>> CS_REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:79
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:662
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State REPORTING going to sleep
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_state_machine.c:387
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1182 Send signal
>> sofia/internal/1000 at XX.XXX.XX.149:5080 [BREAK]
>> 2012-04-27 12:57:24.973781 [DEBUG] switch_core_session.c:1382 Session 204
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Locked, Waiting on external
>> entities
>> 2012-04-27 12:57:24.973781 [NOTICE] switch_core_session.c:1400 Session
>> 204 (sofia/internal/1000 at XX.XXX.XX.149:5080) Ended
>> 2012-04-27 12:57:24.995020 [NOTICE] switch_core_session.c:1402 Close
>> Channel sofia/internal/1000 at XX.XXX.XX.149:5080 [CS_DESTROY]
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:491
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Callstate Change HANGUP -> DOWN
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:494
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) Running State Change CS_DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] mod_sofia.c:374
>> sofia/internal/1000 at XX.XXX.XX.149:5080 SOFIA DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:86
>> sofia/internal/1000 at XX.XXX.XX.149:5080 Standard DESTROY
>> 2012-04-27 12:57:24.995020 [DEBUG] switch_core_state_machine.c:504
>> (sofia/internal/1000 at XX.XXX.XX.149:5080) State DESTROY going to sleep
>>
>>
>> Alfred
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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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From th982a at googlemail.com  Sat Apr 28 21:55:27 2012
From: th982a at googlemail.com (Tamer Higazi)
Date: Sat, 28 Apr 2012 19:55:27 +0200
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
Message-ID: <4F9C2F0F.9090008@googlemail.com>

Hi people!
I am on gentoo, and fetched through it's overlay with git the snapshot
and tried to build FS with mod_gsmopen. During built process it stoped
with this error:


making all mod_gsmopen
copying header files
quiet_libtool: link: warning:
`/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la'
seems to be moved
Compiling gsmopen_protocol.cpp...
In file included from gsmopen_protocol.cpp:1:0:
gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder
Verzeichnis nicht gefunden
Kompilierung beendet.
make[5]: *** [gsmopen_protocol.o] Fehler 1
make[4]: *** [all] Fehler 1
make[3]: *** [mod_gsmopen-all] Fehler 1
make[3]: *** Warte auf noch nicht beendete Prozesse...



Any advises how to solve that?! Would be really cool to connect the cell
through bluetooth and route it to my A100 board.



Tamer


From gmaruzz at gmail.com  Sat Apr 28 21:59:36 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 28 Apr 2012 19:59:36 +0200
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
In-Reply-To: <4F9C2F0F.9090008@googlemail.com>
References: <4F9C2F0F.9090008@googlemail.com>
Message-ID: <CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>

On Sat, Apr 28, 2012 at 7:55 PM, Tamer Higazi <th982a at googlemail.com> wrote:
> Hi people!
> I am on gentoo, and fetched through it's overlay with git the snapshot
> and tried to build FS with mod_gsmopen. During built process it stoped
> with this error:
>
>
> making all mod_gsmopen
> copying header files
> quiet_libtool: link: warning:
> `/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la'
> seems to be moved
> Compiling gsmopen_protocol.cpp...
> In file included from gsmopen_protocol.cpp:1:0:
> gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder


building it:

http://wiki.freeswitch.org/wiki/GSMopen#Building


BTW: it has nothing to do with bluetooth:

http://wiki.freeswitch.org/wiki/GSMopen#What_Is_GSMopen


-giovanni


> Verzeichnis nicht gefunden
> Kompilierung beendet.
> make[5]: *** [gsmopen_protocol.o] Fehler 1
> make[4]: *** [all] Fehler 1
> make[3]: *** [mod_gsmopen-all] Fehler 1
> make[3]: *** Warte auf noch nicht beendete Prozesse...
>
>
>
> Any advises how to solve that?! Would be really cool to connect the cell
> through bluetooth and route it to my A100 board.
>
>
>
> Tamer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anton.jugatsu at gmail.com  Sat Apr 28 22:21:14 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Sat, 28 Apr 2012 22:21:14 +0400
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
In-Reply-To: <CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>
References: <4F9C2F0F.9090008@googlemail.com>
	<CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>
Message-ID: <CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>

Giovanni, hi. Pardon me, but can you checkout a few issues that I reported
at jira about gsmopen.
28.04.2012 22:01 ???????????? "Giovanni Maruzzelli" <gmaruzz at gmail.com>
???????:

> On Sat, Apr 28, 2012 at 7:55 PM, Tamer Higazi <th982a at googlemail.com>
> wrote:
> > Hi people!
> > I am on gentoo, and fetched through it's overlay with git the snapshot
> > and tried to build FS with mod_gsmopen. During built process it stoped
> > with this error:
> >
> >
> > making all mod_gsmopen
> > copying header files
> > quiet_libtool: link: warning:
> > `/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la'
> > seems to be moved
> > Compiling gsmopen_protocol.cpp...
> > In file included from gsmopen_protocol.cpp:1:0:
> > gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder
>
>
> building it:
>
> http://wiki.freeswitch.org/wiki/GSMopen#Building
>
>
> BTW: it has nothing to do with bluetooth:
>
> http://wiki.freeswitch.org/wiki/GSMopen#What_Is_GSMopen
>
>
> -giovanni
>
>
> > Verzeichnis nicht gefunden
> > Kompilierung beendet.
> > make[5]: *** [gsmopen_protocol.o] Fehler 1
> > make[4]: *** [all] Fehler 1
> > make[3]: *** [mod_gsmopen-all] Fehler 1
> > make[3]: *** Warte auf noch nicht beendete Prozesse...
> >
> >
> >
> > Any advises how to solve that?! Would be really cool to connect the cell
> > through bluetooth and route it to my A100 board.
> >
> >
> >
> > Tamer
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From curriegrad2004 at gmail.com  Sat Apr 28 22:36:05 2012
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 28 Apr 2012 11:36:05 -0700
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
In-Reply-To: <CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
References: <4F9C2F0F.9090008@googlemail.com>
	<CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>
	<CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
Message-ID: <CAMWiMy8spaBx2UTpaK5EaaWCqEtZNpwq3bph1WGNFji13iJN-g@mail.gmail.com>

It would also be beneficial if the compiler output was in English.

2012/4/28 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> Giovanni, hi. Pardon me, but can you checkout a few issues that I reported
> at jira about gsmopen.
>
> 28.04.2012 22:01 ???????????? "Giovanni Maruzzelli" <gmaruzz at gmail.com>
> ???????:
>
>> On Sat, Apr 28, 2012 at 7:55 PM, Tamer Higazi <th982a at googlemail.com>
>> wrote:
>> > Hi people!
>> > I am on gentoo, and fetched through it's overlay with git the snapshot
>> > and tried to build FS with mod_gsmopen. During built process it stoped
>> > with this error:
>> >
>> >
>> > making all mod_gsmopen
>> > copying header files
>> > quiet_libtool: link: warning:
>> > `/usr/lib/gcc/x86_64-pc-linux-gnu/4.5.3/../../../../lib64/libodbc.la'
>> > seems to be moved
>> > Compiling gsmopen_protocol.cpp...
>> > In file included from gsmopen_protocol.cpp:1:0:
>> > gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h: Datei oder
>>
>>
>> building it:
>>
>> http://wiki.freeswitch.org/wiki/GSMopen#Building
>>
>>
>> BTW: it has nothing to do with bluetooth:
>>
>> http://wiki.freeswitch.org/wiki/GSMopen#What_Is_GSMopen
>>
>>
>> -giovanni
>>
>>
>> > Verzeichnis nicht gefunden
>> > Kompilierung beendet.
>> > make[5]: *** [gsmopen_protocol.o] Fehler 1
>> > make[4]: *** [all] Fehler 1
>> > make[3]: *** [mod_gsmopen-all] Fehler 1
>> > make[3]: *** Warte auf noch nicht beendete Prozesse...
>> >
>> >
>> >
>> > Any advises how to solve that?! Would be really cool to connect the cell
>> > through bluetooth and route it to my A100 board.
>> >
>> >
>> >
>> > Tamer
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From lazyvirus at gmx.com  Sat Apr 28 22:46:12 2012
From: lazyvirus at gmx.com (Bzzz)
Date: Sat, 28 Apr 2012 20:46:12 +0200
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
In-Reply-To: <CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
References: <4F9C2F0F.9090008@googlemail.com>
	<CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>
	<CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
Message-ID: <20120428204612.7581f2f2@anubis.defcon1>

On Sat, 28 Apr 2012 22:21:14 +0400
Anton Kvashenkin <anton.jugatsu at gmail.com> wrote:

> > > gsmopen.h:121:26: schwerwiegender Fehler: ctb-0.16/ctb.h:
> > > Datei oder

as currierad said, english messages would be better; issue an
export LC_MESSAGES=C in the same console and relaunch compilation.

-- 
In Greene, New York, it is illegal to eat peanuts and walk backwards
on the sidewalks when a concert is on.


From gmaruzz at gmail.com  Sat Apr 28 22:57:36 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 28 Apr 2012 20:57:36 +0200
Subject: [Freeswitch-users] can't build mod_gsmopen ?!
In-Reply-To: <CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
References: <4F9C2F0F.9090008@googlemail.com>
	<CALXCt0oq5FYie37q5jBJ=GqENU_RgWCefzpRFWF=6EmioSSK3w@mail.gmail.com>
	<CANBhyPB3Bh=FUt2_=cG83prvMyBR7vT+8xOm4c9Kqx3DyJumfA@mail.gmail.com>
Message-ID: <CALXCt0q6TWhzbwa75mHcTDu9gD5VzqHg7c9=B0XHKmZEzE0KZA@mail.gmail.com>

2012/4/28 Anton Kvashenkin <anton.jugatsu at gmail.com>:
> Giovanni, hi. Pardon me, but can you checkout a few issues that I reported
> at jira about gsmopen.

I'll fix it as soon as I have time to.
Maybe tomorrow. Maybe.


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From bdfoster at endigotech.com  Sat Apr 28 23:16:52 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 28 Apr 2012 15:16:52 -0400
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
 not reachable/registered
In-Reply-To: <CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
	<4F9B1CDE.6040509@quentustech.com>
	<CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>
	<CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>
Message-ID: <CAKV-jFxH8xD=C9ntqMxEm958bJ_oO6W440LgfGEOsPjUSMTztw@mail.gmail.com>

As was mentioned prior, take a look at the continue_on_fail variable, and
you can come up with some slick conditions to fill in that void. It should
be fairly straightforward.

-BDF

On Sat, Apr 28, 2012 at 9:46 AM, Travis Langhals <tlanghals at gmail.com>wrote:

> Thanks for the info, I have the voicemail action working but if I add a
> second bridge command between my extension bridge and voicemail actions it
> rings the cell phone if the extension doesn't answer within my 20s timeout.
>  I'm trying to make it only ring the cell phone if the extension is not
> online and then go to voicemail.  If the extension is online it should just
> ring the extension and then go to voicemail.
>
>
> On Fri, Apr 27, 2012 at 9:39 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Look at the local exensions dialplan in default.xml for an example,
>> specifically how it is forwarded on to voicemail. You could get slick and
>> do conditions and set a param inside of the users .xml file in the
>> directory to tell the dialplan where to try next. Also look at the
>> call_timeout variable.
>>
>> -BDF
>> On Apr 27, 2012 6:26 PM, "William King" <william.king at quentustech.com>
>> wrote:
>>
>>>  You want to do research into the continue-on-fail  and the variable
>>> hangup after bridge.
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>
>>> Basically you can have multiple bridge actions in an extension and the
>>> first to successfully bridge a call will be the last action the call will
>>> make.
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518william.king at quentustech.com
>>>
>>>
>>> On 04/27/2012 03:01 PM, Travis Langhals wrote:
>>>
>>> Could someone please help with an example dialplan for call failover to
>>> an external DID (i.e. cell phone or pots line) in case the the user has a
>>> network failure can cannot be reached?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
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From bdfoster at endigotech.com  Sun Apr 29 01:18:10 2012
From: bdfoster at endigotech.com (Brian Foster)
Date: Sat, 28 Apr 2012 17:18:10 -0400
Subject: [Freeswitch-users] rfc2833 first Marker Bit not set to True on
	bleg
In-Reply-To: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>
References: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>
Message-ID: <CAKV-jFzFvCCjCM6_z7np5FYUc1_Hr4Q__=ga05w5+VXKngsERA@mail.gmail.com>

Please report bugs to JIRA. The mailing list is not the place for reporting
bugs.

Thread closed.

-BDF
On Apr 28, 2012 5:54 AM, "Bernard Murphy" <bernard.david.murphy at gmail.com>
wrote:

> On the very latest GIT and previous , Freeswitch doesn't seem to set the
> first Marker Bit to true on an rfc2833 event on the bleg of a call out to a
> gateway
>
> on the A-leg the first event from my UA to Freeswitch is correct with
> duration as 0 and Marker bit set to True. Then the second event increments
> the duration and Market bit is set to false.
> but on the B-leg from freeswitch to the cisco gateway the first event has
> duration 80 (may or may not be a problem) but the Marker bit is set to
> False which must be wrong ? Duration then increments and Marker bit remains
> false.
>
> I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/> to
> make the duration start at zero but it seemed to have no effect.
>
>
> *ALEG from SIP UA to FREESWITCH*
> No.     Time Source                Destination           Protocol Length
> Info
>    5254 23   10.0.47.110           10.50.47.30           RTP EVENT 60
> Payload type=RTP Event, DTMF One 1
>
> Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
> 10.50.47.30 (10.50.47.30)
> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1122)]
>         [Setup frame: 1122]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     1... .... = Marker: True
>     Payload type: telephone-event (101)
>     Sequence number: 26364
>     [Extended sequence number: 91900]
>     Timestamp: 133920
>     Synchronization Source identifier: 0x03bf0676 (62850678)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 1010 = Volume: 10
>     Event Duration: 0
>
> *ALEG SECOND EVENT FROM SIP UA TO FREESWITCH*
>
> No.     Time Source                Destination           Protocol Length
> Info
>    5255 23   10.0.47.110           10.50.47.30           RTP EVENT 60
> Payload type=RTP Event, DTMF One 1
>
> Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
> 10.50.47.30 (10.50.47.30)
> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1122)]
>         [Setup frame: 1122]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 26365
>     [Extended sequence number: 91901]
>     Timestamp: 133920
>     Synchronization Source identifier: 0x03bf0676 (62850678)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 1010 = Volume: 10
>     Event Duration: 160
>
> *BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO*
> No.     Time Source                Destination           Protocol Length
> Info
>    5329 23   10.50.47.30           10.50.47.19           RTP EVENT 58
> Payload type=RTP Event, DTMF One 1
>
> Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
> 10.50.47.19 (10.50.47.19)
> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1113)]
>         [Setup frame: 1113]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 24716
>     [Extended sequence number: 90252]
>     Timestamp: 158240
>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 0111 = Volume: 7
>     Event Duration: 80
>
> *BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO*
> No.     Time Source                Destination           Protocol Length
> Info
>    5331 23   10.50.47.30           10.50.47.19           RTP EVENT 58
> Payload type=RTP Event, DTMF One 1
>
> Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
> 10.50.47.19 (10.50.47.19)
> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1113)]
>         [Setup frame: 1113]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 24717
>     [Extended sequence number: 90253]
>     Timestamp: 158240
>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 0111 = Volume: 7
>     Event Duration: 160
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From avlubimov at gmail.com  Sun Apr 29 06:41:20 2012
From: avlubimov at gmail.com (=?KOI8-R?B?4czFy9PFyiDswMLJzc/X?=)
Date: Sun, 29 Apr 2012 06:41:20 +0400
Subject: [Freeswitch-users] wrong detection dtmf rfc2833
In-Reply-To: <SNT127-W16680B02C8CBF4363B6079D5250@phx.gbl>
References: <4F9AADB5.9060200@gmail.com>
	<CAA_857vSoH-1BC8QbsyD=XJ0F2nf30qwBvkw5OGBt=PvbMPiPQ@mail.gmail.com>
	<SNT127-W16680B02C8CBF4363B6079D5250@phx.gbl>
Message-ID: <CAA_857sdN=i=qqdcg+GANZxAXk=yzn-=3RvByXrfdRqXLxKThw@mail.gmail.com>

2012/4/27 Yuriy Nasida <nasida at live.ru>

>  it depend from your  incoming sip provider as well.
> did you try to detect dtmf  for incoming calls  in media stream (inband)
>  instead of using rfc2833 ?
> <param name="dtmf-type" value="none"/>  - on  incoming  sip profile
> <action application="start_dtmf" /> - in dialplan before bridge
>
>
Yes, you are right.

I was found mistake in my dialplan.

After  remove  <action application="start_dtmf" /> from dialplan ( before
start ivr), now, dtmf recognized 100% correctly.

Thanks!
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From brian at freeswitch.org  Sun Apr 29 07:36:54 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 28 Apr 2012 22:36:54 -0500
Subject: [Freeswitch-users] rfc2833 first Marker Bit not set to True on
	bleg
In-Reply-To: <CAKV-jFzFvCCjCM6_z7np5FYUc1_Hr4Q__=ga05w5+VXKngsERA@mail.gmail.com>
References: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>
	<CAKV-jFzFvCCjCM6_z7np5FYUc1_Hr4Q__=ga05w5+VXKngsERA@mail.gmail.com>
Message-ID: <2825583103416496097@unknownmsgid>

Not a bug!  Set rtp bugs param to clear!

/b

Sent from my iPhone

On Apr 28, 2012, at 4:19 PM, Brian Foster <bdfoster at endigotech.com> wrote:

Please report bugs to JIRA. The mailing list is not the place for reporting
bugs.

Thread closed.

-BDF
On Apr 28, 2012 5:54 AM, "Bernard Murphy" <bernard.david.murphy at gmail.com>
wrote:

> On the very latest GIT and previous , Freeswitch doesn't seem to set the
> first Marker Bit to true on an rfc2833 event on the bleg of a call out to a
> gateway
>
> on the A-leg the first event from my UA to Freeswitch is correct with
> duration as 0 and Marker bit set to True. Then the second event increments
> the duration and Market bit is set to false.
> but on the B-leg from freeswitch to the cisco gateway the first event has
> duration 80 (may or may not be a problem) but the Marker bit is set to
> False which must be wrong ? Duration then increments and Marker bit remains
> false.
>
> I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/> to
> make the duration start at zero but it seemed to have no effect.
>
>
> *ALEG from SIP UA to FREESWITCH*
> No.     Time Source                Destination           Protocol Length
> Info
>    5254 23   10.0.47.110           10.50.47.30           RTP EVENT 60
> Payload type=RTP Event, DTMF One 1
>
> Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
> 10.50.47.30 (10.50.47.30)
> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1122)]
>         [Setup frame: 1122]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     1... .... = Marker: True
>     Payload type: telephone-event (101)
>     Sequence number: 26364
>     [Extended sequence number: 91900]
>     Timestamp: 133920
>     Synchronization Source identifier: 0x03bf0676 (62850678)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 1010 = Volume: 10
>     Event Duration: 0
>
> *ALEG SECOND EVENT FROM SIP UA TO FREESWITCH*
>
> No.     Time Source                Destination           Protocol Length
> Info
>    5255 23   10.0.47.110           10.50.47.30           RTP EVENT 60
> Payload type=RTP Event, DTMF One 1
>
> Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
> 10.50.47.30 (10.50.47.30)
> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1122)]
>         [Setup frame: 1122]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 26365
>     [Extended sequence number: 91901]
>     Timestamp: 133920
>     Synchronization Source identifier: 0x03bf0676 (62850678)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 1010 = Volume: 10
>     Event Duration: 160
>
> *BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO*
> No.     Time Source                Destination           Protocol Length
> Info
>    5329 23   10.50.47.30           10.50.47.19           RTP EVENT 58
> Payload type=RTP Event, DTMF One 1
>
> Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
> 10.50.47.19 (10.50.47.19)
> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1113)]
>         [Setup frame: 1113]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 24716
>     [Extended sequence number: 90252]
>     Timestamp: 158240
>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 0111 = Volume: 7
>     Event Duration: 80
>
> *BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO*
> No.     Time Source                Destination           Protocol Length
> Info
>    5331 23   10.50.47.30           10.50.47.19           RTP EVENT 58
> Payload type=RTP Event, DTMF One 1
>
> Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
> 10.50.47.19 (10.50.47.19)
> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
> Real-Time Transport Protocol
>     [Stream setup by SDP (frame 1113)]
>         [Setup frame: 1113]
>         [Setup Method: SDP]
>     10.. .... = Version: RFC 1889 Version (2)
>     ..0. .... = Padding: False
>     ...0 .... = Extension: False
>     .... 0000 = Contributing source identifiers count: 0
>     0... .... = Marker: False
>     Payload type: telephone-event (101)
>     Sequence number: 24717
>     [Extended sequence number: 90253]
>     Timestamp: 158240
>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
> RFC 2833 RTP Event
>     Event ID: DTMF One 1 (1)
>     0... .... = End of Event: False
>     .0.. .... = Reserved: False
>     ..00 0111 = Volume: 7
>     Event Duration: 160
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Sun Apr 29 07:37:45 2012
From: brian at freeswitch.org (Brian West)
Date: Sat, 28 Apr 2012 22:37:45 -0500
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
 not reachable/registered
In-Reply-To: <CAKV-jFxH8xD=C9ntqMxEm958bJ_oO6W440LgfGEOsPjUSMTztw@mail.gmail.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
	<4F9B1CDE.6040509@quentustech.com>
	<CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>
	<CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>
	<CAKV-jFxH8xD=C9ntqMxEm958bJ_oO6W440LgfGEOsPjUSMTztw@mail.gmail.com>
Message-ID: <-3302707391787771903@unknownmsgid>

Or use XML Curl and you know this before you even generate the dial plan!

/b

Sent from my iPhone

On Apr 28, 2012, at 2:18 PM, Brian Foster <bdfoster at endigotech.com> wrote:

As was mentioned prior, take a look at the continue_on_fail variable, and
you can come up with some slick conditions to fill in that void. It should
be fairly straightforward.

-BDF

On Sat, Apr 28, 2012 at 9:46 AM, Travis Langhals <tlanghals at gmail.com>wrote:

> Thanks for the info, I have the voicemail action working but if I add a
> second bridge command between my extension bridge and voicemail actions it
> rings the cell phone if the extension doesn't answer within my 20s timeout.
>  I'm trying to make it only ring the cell phone if the extension is not
> online and then go to voicemail.  If the extension is online it should just
> ring the extension and then go to voicemail.
>
>
> On Fri, Apr 27, 2012 at 9:39 PM, Brian Foster <bdfoster at endigotech.com>wrote:
>
>> Look at the local exensions dialplan in default.xml for an example,
>> specifically how it is forwarded on to voicemail. You could get slick and
>> do conditions and set a param inside of the users .xml file in the
>> directory to tell the dialplan where to try next. Also look at the
>> call_timeout variable.
>>
>> -BDF
>> On Apr 27, 2012 6:26 PM, "William King" <william.king at quentustech.com>
>> wrote:
>>
>>>  You want to do research into the continue-on-fail  and the variable
>>> hangup after bridge.
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>
>>> Basically you can have multiple bridge actions in an extension and the
>>> first to successfully bridge a call will be the last action the call will
>>> make.
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518william.king at quentustech.com
>>>
>>>
>>> On 04/27/2012 03:01 PM, Travis Langhals wrote:
>>>
>>> Could someone please help with an example dialplan for call failover to
>>> an external DID (i.e. cell phone or pots line) in case the the user has a
>>> network failure can cannot be reached?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Brian D. Foster
Endigo Computer LLC
Email: bdfoster at endigotech.com
Phone: 317-800-7876
Indianapolis, Indiana, USA

This message contains confidential information and is intended for those
listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If
you are not the intended recipient you are notified that disclosing,
copying, distributing or taking any action in reliance on the contents of
this information is strictly prohibited. E-mail transmission cannot be
guaranteed to be secure or error-free as information could be intercepted,
corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.
The sender therefore does not accept liability for any errors or omissions
in the contents of this message, which arise as a result of e-mail
transmission. If verification is required please request a hard-copy
version.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From a.afzali2003 at gmail.com  Sun Apr 29 09:05:14 2012
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 29 Apr 2012 09:35:14 +0430
Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch
	to an offline agent
Message-ID: <CABK0fC4WibNuH9VEEm0sGK0hfQckZGKpD4OzHr-FsRce7u+Z8A@mail.gmail.com>

Dear FreeSWITCHers,

The offline agents need to make outgoing calls while waiting for call
arrivals. To prevent an influence outgoing by incoming call, they need
to logout, making call and then login to the queue ! also doing logout &
login automatically alters the actual login time :(
I'm looking for a solution which can disable call dispatch temporary for a
specific agent.
Appreciate all comments,
-- afshin
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From bernard.david.murphy at gmail.com  Sun Apr 29 14:17:01 2012
From: bernard.david.murphy at gmail.com (Bernard Murphy)
Date: Sun, 29 Apr 2012 11:17:01 +0100
Subject: [Freeswitch-users] rfc2833 first Marker Bit not set to True on
	bleg
In-Reply-To: <2825583103416496097@unknownmsgid>
References: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>
	<CAKV-jFzFvCCjCM6_z7np5FYUc1_Hr4Q__=ga05w5+VXKngsERA@mail.gmail.com>
	<2825583103416496097@unknownmsgid>
Message-ID: <CAOte9J5PvrAazPTP00Pf6LLNh+uLgLxxA7qsncN-AQUC2_xYhQ@mail.gmail.com>

Thank-you, <param name="auto-rtp-bugs" value="clear"/> does indeed fix the
problem of Bleg, first bit Marker not set to True for a Cisco 2901 SIP
Gateway IOS 15.0(1r)M12

I appreciate the assistance from the group as ever.

Regards
Bernie


On Sun, Apr 29, 2012 at 4:36 AM, Brian West <brian at freeswitch.org> wrote:

> Not a bug!  Set rtp bugs param to clear!
>
> /b
>
> Sent from my iPhone
>
> On Apr 28, 2012, at 4:19 PM, Brian Foster <bdfoster at endigotech.com> wrote:
>
> Please report bugs to JIRA. The mailing list is not the place for
> reporting bugs.
>
> Thread closed.
>
> -BDF
> On Apr 28, 2012 5:54 AM, "Bernard Murphy" <bernard.david.murphy at gmail.com>
> wrote:
>
>> On the very latest GIT and previous , Freeswitch doesn't seem to set the
>> first Marker Bit to true on an rfc2833 event on the bleg of a call out to a
>> gateway
>>
>> on the A-leg the first event from my UA to Freeswitch is correct with
>> duration as 0 and Marker bit set to True. Then the second event increments
>> the duration and Market bit is set to false.
>> but on the B-leg from freeswitch to the cisco gateway the first event has
>> duration 80 (may or may not be a problem) but the Marker bit is set to
>> False which must be wrong ? Duration then increments and Marker bit remains
>> false.
>>
>> I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/> to
>> make the duration start at zero but it seemed to have no effect.
>>
>>
>> *ALEG from SIP UA to FREESWITCH*
>> No.     Time Source                Destination           Protocol Length
>> Info
>>    5254 23   10.0.47.110           10.50.47.30           RTP EVENT 60
>> Payload type=RTP Event, DTMF One 1
>>
>> Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
>> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
>> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
>> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
>> 10.50.47.30 (10.50.47.30)
>> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
>> Real-Time Transport Protocol
>>     [Stream setup by SDP (frame 1122)]
>>         [Setup frame: 1122]
>>         [Setup Method: SDP]
>>     10.. .... = Version: RFC 1889 Version (2)
>>     ..0. .... = Padding: False
>>     ...0 .... = Extension: False
>>     .... 0000 = Contributing source identifiers count: 0
>>     1... .... = Marker: True
>>     Payload type: telephone-event (101)
>>     Sequence number: 26364
>>     [Extended sequence number: 91900]
>>     Timestamp: 133920
>>     Synchronization Source identifier: 0x03bf0676 (62850678)
>> RFC 2833 RTP Event
>>     Event ID: DTMF One 1 (1)
>>     0... .... = End of Event: False
>>     .0.. .... = Reserved: False
>>     ..00 1010 = Volume: 10
>>     Event Duration: 0
>>
>> *ALEG SECOND EVENT FROM SIP UA TO FREESWITCH*
>>
>> No.     Time Source                Destination           Protocol Length
>> Info
>>    5255 23   10.0.47.110           10.50.47.30           RTP EVENT 60
>> Payload type=RTP Event, DTMF One 1
>>
>> Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
>> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
>> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
>> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
>> 10.50.47.30 (10.50.47.30)
>> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
>> Real-Time Transport Protocol
>>     [Stream setup by SDP (frame 1122)]
>>         [Setup frame: 1122]
>>         [Setup Method: SDP]
>>     10.. .... = Version: RFC 1889 Version (2)
>>     ..0. .... = Padding: False
>>     ...0 .... = Extension: False
>>     .... 0000 = Contributing source identifiers count: 0
>>     0... .... = Marker: False
>>     Payload type: telephone-event (101)
>>     Sequence number: 26365
>>     [Extended sequence number: 91901]
>>     Timestamp: 133920
>>     Synchronization Source identifier: 0x03bf0676 (62850678)
>> RFC 2833 RTP Event
>>     Event ID: DTMF One 1 (1)
>>     0... .... = End of Event: False
>>     .0.. .... = Reserved: False
>>     ..00 1010 = Volume: 10
>>     Event Duration: 160
>>
>> *BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO*
>> No.     Time Source                Destination           Protocol Length
>> Info
>>    5329 23   10.50.47.30           10.50.47.19           RTP EVENT 58
>> Payload type=RTP Event, DTMF One 1
>>
>> Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
>> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
>> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
>> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
>> 10.50.47.19 (10.50.47.19)
>> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
>> Real-Time Transport Protocol
>>     [Stream setup by SDP (frame 1113)]
>>         [Setup frame: 1113]
>>         [Setup Method: SDP]
>>     10.. .... = Version: RFC 1889 Version (2)
>>     ..0. .... = Padding: False
>>     ...0 .... = Extension: False
>>     .... 0000 = Contributing source identifiers count: 0
>>     0... .... = Marker: False
>>     Payload type: telephone-event (101)
>>     Sequence number: 24716
>>     [Extended sequence number: 90252]
>>     Timestamp: 158240
>>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
>> RFC 2833 RTP Event
>>     Event ID: DTMF One 1 (1)
>>     0... .... = End of Event: False
>>     .0.. .... = Reserved: False
>>     ..00 0111 = Volume: 7
>>     Event Duration: 80
>>
>> *BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO*
>> No.     Time Source                Destination           Protocol Length
>> Info
>>    5331 23   10.50.47.30           10.50.47.19           RTP EVENT 58
>> Payload type=RTP Event, DTMF One 1
>>
>> Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
>> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
>> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
>> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
>> 10.50.47.19 (10.50.47.19)
>> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
>> Real-Time Transport Protocol
>>     [Stream setup by SDP (frame 1113)]
>>         [Setup frame: 1113]
>>         [Setup Method: SDP]
>>     10.. .... = Version: RFC 1889 Version (2)
>>     ..0. .... = Padding: False
>>     ...0 .... = Extension: False
>>     .... 0000 = Contributing source identifiers count: 0
>>     0... .... = Marker: False
>>     Payload type: telephone-event (101)
>>     Sequence number: 24717
>>     [Extended sequence number: 90253]
>>     Timestamp: 158240
>>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
>> RFC 2833 RTP Event
>>     Event ID: DTMF One 1 (1)
>>     0... .... = End of Event: False
>>     .0.. .... = Reserved: False
>>     ..00 0111 = Volume: 7
>>     Event Duration: 160
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From oseslija at gmail.com  Sun Apr 29 17:32:10 2012
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sun, 29 Apr 2012 15:32:10 +0200
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <4F9AC3F7.1000601@communicatefreely.net>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
	<CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>
	<B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>
	<CAKV-jFydnHBYZkrPVxS8Py6yRijThc6NF_f54mEVpAUHhuEuSw@mail.gmail.com>
	<4F9AC3F7.1000601@communicatefreely.net>
Message-ID: <CAM21H-PdJOFmFpmRSFv=d56w5ae0dS-LfMfQpomNhfTudbHu1w@mail.gmail.com>

Hi,

You can set user at domain for BLFs on Linksys SPA 9xx and Cisco SPA 5xx
phones.

On Fri, Apr 27, 2012 at 6:06 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Hi Rick,
>
> Freeswitch has no problem with this at all, but finding a phone that can
> do it may be the issue.
>
> You would need to find a phone that can configure BLFs using a specified
> SIP domain, not just the SIP username.
>
> For example, on an Aastra phone (it's what I know), you configure the
> SIP domain that is associated with a given registration.  When you
> create BLF keys, you only specify the username - the domain that is
> subscribed to is whatever the domain is for that registration.
>
> Somehow, you need to map a button and corresponding subscription to each
> receptionist phone at their respective domain, and this may not be easy.
>
> Off the top of my head, I don't know which phones can do this.  Polycom
> might be able to do it, as they seem to be better at making the full SIP
> URI accessible to the user.  I know it's possible to dial a SIP URI
> directly on a Polycom, whereas Aastra phones have to have it in the
> speed dial list.
>
> Last I checked, there was some sort of option to only let people
> subscribe to their own domain, but I think it is off by default.
>
> Another thing you will need to check, is the "force-subsription-domain"
> and "force-subscription-db-domain" (sic) in your SIP profile to make
> sure that the domain you subscribe to is what is actually used in the
> subscription.
>
> -Tim
>
> Brian Foster wrote:
> >
> >
> > On Tue, Apr 24, 2012 at 12:43 AM, Rick Guyton
> > <Rick at claritycomputer.com <mailto:Rick at claritycomputer.com>> wrote:
> >
> >     To be clear, I?m in utter amazement of what freeswitch can do and
> >     appreciate everyone who has contributed to making such an awesome
> >     application.
> >
> >
> > Welcome to the club ;-)
> >
> >
> >     So, when I can?t do something, I assume it is because of my own
> >     stupidity. I had assumed that I needed to make a ?permissions?
> >     change somewhere or add the users from other domains as an alias
> >     or some sort. If it can?t be done, that makes sense. I meant no
> >     disrespect to Tony or anyone else who has contributed. I would
> >     just put them all in the same domain, but I must separate the
> >     companies for mostly political reasons.
> >
> >
> > Understandable.
> >
> > Depending on your setup, you might just be able to do this on the
> > phones themselves. I don't know if you use these, but you know the
> > little attendant consoles you can plug into phones like polycom and
> > aastra? Can you register to the lines you want to monitor with those?
> > I've never actually had one (I'd like to get one for testing) but I'm
> > fairly certain you can. That would also depend on your network setup.
> > If you've got all the companies on their own LAN's then you're screwed
> > unless freeswitch can be fully accessed on each of those LAN's. Might
> > be something to check out.
> >
> >     * *
> >
> > -BDF
> >
> >
> >
> >     *From:* freeswitch-users-bounces at lists.freeswitch.org
> >     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> >     [mailto:freeswitch-users-bounces at lists.freeswitch.org
> >     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf
> >     Of *Chris Chen
> >     *Sent:* Monday, April 23, 2012 6:30 PM
> >     *To:* FreeSWITCH Users Help
> >     *Subject:* Re: [Freeswitch-users] BLF in Multi-tenant config
> >
> >
> >
> >     Agree with BDF.
> >
> >
> >
> >     By design, in a multi-tenant environment, 101 at companyA should only
> >     have BLF status for any extension within companyA,  otherwise why
> >     should you have multi-tenant setup, just have them all in the same
> >     domain. We should really appreciate Tony crew having built the FS
> >     to be really capable of providing multi-tenant functions without
> >     too many tweaks.
> >
> >
> >
> >     Just my two cents.
> >
> >
> >
> >     Chris
> >
> >
> >
> >     On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster
> >     <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>> wrote:
> >
> >     I'm not sure that multi-tenant wouldn't support that. I think the
> >     point of multi-tenant is to keep everything seperated. There might
> >     be a hack for this, though. Nothing comes to mind for me.
> >
> >     -BDF
> >
> >     On Apr 23, 2012 7:31 PM, "Rick Guyton" <Rick at claritycomputer.com
> >     <mailto:Rick at claritycomputer.com>> wrote:
> >
> >         We have 3 different companies in this office and we all share
> >         the same freeswitch server, I?ll call then CompanyA, CompanyB
> >         and CompanyC. CompanyA, B and C all employ their own
> >         receptionists, and they are all assigned extension 101 on in
> >         their respective domains. However, the receptionists all work
> >         very tightly together and basically share work between them.
> >         So, I need to assign the BLF status of 101 at CompanyA,
> >         101 at CompanyB and 101 at CompanyC to indicator lights across all
> >         phones in the building. My problem is that users in CompanyA
> >         can?t see the status of 101 at CompanyB or 101 at CompanyC. Users in
> >         CompanyB can?t see 101 at CompanyA or 101 at CompanyC and so on?
> >
> >
> >
> >         So, my question is, how do I allow users of one domain in a
> >         multi-tenant configuration to see the status of other users in
> >         a different domain?
> >
> >
> >
> >
> _________________________________________________________________________
> >         Professional FreeSWITCH Consulting Services:
> >         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >         http://www.freeswitchsolutions.com
> >
> >         
> >         
> >
> >         Official FreeSWITCH Sites
> >         http://www.freeswitch.org
> >         http://wiki.freeswitch.org
> >         http://www.cluecon.com
> >
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> >
> >
> ------------------------------------------------------------------------
> >
> >     If this email is spam, report it to www.OnlyMyEmail.com
> >     <
> https://support.onlymyemail.com/view/report_spam/MTM3Mjc1OjE0MTI0MDQ4MjI6cmlja0BjbGFyaXR5Y29tcHV0ZXIuY29tOmRlbGl2ZXJlZA
> >
> >
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Brian D. Foster
> > Endigo Computer LLC
> > Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
> > Phone: 317-800-7876
> > Indianapolis, Indiana, USA
> >
> > This message contains confidential information and is intended for
> > those listed in the "To:", "CC:", and/or "BCC:" fields of the message
> > header. If you are not the intended recipient you are notified that
> > disclosing, copying, distributing or taking any action in reliance on
> > the contents of this information is strictly prohibited. E-mail
> > transmission cannot be guaranteed to be secure or error-free as
> > information could be intercepted, corrupted, lost, destroyed, arrive
> > late or incomplete, or contain viruses. The sender therefore does not
> > accept liability for any errors or omissions in the contents of this
> > message, which arise as a result of e-mail transmission. If
> > verification is required please request a hard-copy version.
> >
> > ------------------------------------------------------------------------
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Mon Apr 30 00:21:00 2012
From: abaci64 at gmail.com (Abaci)
Date: Sun, 29 Apr 2012 16:21:00 -0400
Subject: [Freeswitch-users] BLF in Multi-tenant config
In-Reply-To: <CAM21H-PdJOFmFpmRSFv=d56w5ae0dS-LfMfQpomNhfTudbHu1w@mail.gmail.com>
References: <B101E46F26FEC54ABFB2CA9BFBCC38BD82534F8B@SERVER.clarity.local>
	<CAKV-jFyyNV7pPRfeM+dd+5s8GoNLcdx2Ycfw3Tt1swEiynsXBw@mail.gmail.com>
	<CAFEMH7MGWe4ddNSB0PwL728TMCD-1PvS60S=v3kmL3_iSt1qZQ@mail.gmail.com>
	<B101E46F26FEC54ABFB2CA9BFBCC38BD82535522@SERVER.clarity.local>
	<CAKV-jFydnHBYZkrPVxS8Py6yRijThc6NF_f54mEVpAUHhuEuSw@mail.gmail.com>
	<4F9AC3F7.1000601@communicatefreely.net>
	<CAM21H-PdJOFmFpmRSFv=d56w5ae0dS-LfMfQpomNhfTudbHu1w@mail.gmail.com>
Message-ID: <4F9DA2AC.2050400@gmail.com>

So does someone know how to restrict in FreeSWITCH not to allow phones 
to subscribe to another domain? Tim mention there is an option in 
FreeSWITCH but didn't give any details.

On 4/29/2012 9:32 AM, Ognjen Seslija wrote:
> Hi,
>
> You can set user at domain for BLFs on Linksys SPA 9xx and Cisco SPA 5xx 
> phones.
>
> On Fri, Apr 27, 2012 at 6:06 PM, Tim St. Pierre 
> <fs-list at communicatefreely.net <mailto:fs-list at communicatefreely.net>> 
> wrote:
>
>     Hi Rick,
>
>     Freeswitch has no problem with this at all, but finding a phone
>     that can
>     do it may be the issue.
>
>     You would need to find a phone that can configure BLFs using a
>     specified
>     SIP domain, not just the SIP username.
>
>     For example, on an Aastra phone (it's what I know), you configure the
>     SIP domain that is associated with a given registration.  When you
>     create BLF keys, you only specify the username - the domain that is
>     subscribed to is whatever the domain is for that registration.
>
>     Somehow, you need to map a button and corresponding subscription
>     to each
>     receptionist phone at their respective domain, and this may not be
>     easy.
>
>     Off the top of my head, I don't know which phones can do this.
>      Polycom
>     might be able to do it, as they seem to be better at making the
>     full SIP
>     URI accessible to the user.  I know it's possible to dial a SIP URI
>     directly on a Polycom, whereas Aastra phones have to have it in the
>     speed dial list.
>
>     Last I checked, there was some sort of option to only let people
>     subscribe to their own domain, but I think it is off by default.
>
>     Another thing you will need to check, is the
>     "force-subsription-domain"
>     and "force-subscription-db-domain" (sic) in your SIP profile to make
>     sure that the domain you subscribe to is what is actually used in the
>     subscription.
>
>     -Tim
>
>     Brian Foster wrote:
>     >
>     >
>     > On Tue, Apr 24, 2012 at 12:43 AM, Rick Guyton
>     > <Rick at claritycomputer.com <mailto:Rick at claritycomputer.com>
>     <mailto:Rick at claritycomputer.com
>     <mailto:Rick at claritycomputer.com>>> wrote:
>     >
>     >     To be clear, I'm in utter amazement of what freeswitch can
>     do and
>     >     appreciate everyone who has contributed to making such an
>     awesome
>     >     application.
>     >
>     >
>     > Welcome to the club ;-)
>     >
>     >
>     >     So, when I can't do something, I assume it is because of my own
>     >     stupidity. I had assumed that I needed to make a "permissions"
>     >     change somewhere or add the users from other domains as an alias
>     >     or some sort. If it can't be done, that makes sense. I meant no
>     >     disrespect to Tony or anyone else who has contributed. I would
>     >     just put them all in the same domain, but I must separate the
>     >     companies for mostly political reasons.
>     >
>     >
>     > Understandable.
>     >
>     > Depending on your setup, you might just be able to do this on the
>     > phones themselves. I don't know if you use these, but you know the
>     > little attendant consoles you can plug into phones like polycom and
>     > aastra? Can you register to the lines you want to monitor with
>     those?
>     > I've never actually had one (I'd like to get one for testing)
>     but I'm
>     > fairly certain you can. That would also depend on your network
>     setup.
>     > If you've got all the companies on their own LAN's then you're
>     screwed
>     > unless freeswitch can be fully accessed on each of those LAN's.
>     Might
>     > be something to check out.
>     >
>     >     * *
>     >
>     > -BDF
>     >
>     >
>     >
>     >     *From:* freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     > <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     >     [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     > <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>] *On Behalf
>     >     Of *Chris Chen
>     >     *Sent:* Monday, April 23, 2012 6:30 PM
>     >     *To:* FreeSWITCH Users Help
>     >     *Subject:* Re: [Freeswitch-users] BLF in Multi-tenant config
>     >
>     >
>     >
>     >     Agree with BDF.
>     >
>     >
>     >
>     >     By design, in a multi-tenant environment, 101 at companyA
>     should only
>     >     have BLF status for any extension within companyA,
>      otherwise why
>     >     should you have multi-tenant setup, just have them all in
>     the same
>     >     domain. We should really appreciate Tony crew having built
>     the FS
>     >     to be really capable of providing multi-tenant functions without
>     >     too many tweaks.
>     >
>     >
>     >
>     >     Just my two cents.
>     >
>     >
>     >
>     >     Chris
>     >
>     >
>     >
>     >     On Mon, Apr 23, 2012 at 9:21 PM, Brian Foster
>     > <bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>     <mailto:bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>>>
>     wrote:
>     >
>     >     I'm not sure that multi-tenant wouldn't support that. I
>     think the
>     >     point of multi-tenant is to keep everything seperated. There
>     might
>     >     be a hack for this, though. Nothing comes to mind for me.
>     >
>     >     -BDF
>     >
>     >     On Apr 23, 2012 7:31 PM, "Rick Guyton"
>     <Rick at claritycomputer.com <mailto:Rick at claritycomputer.com>
>     > <mailto:Rick at claritycomputer.com
>     <mailto:Rick at claritycomputer.com>>> wrote:
>     >
>     >         We have 3 different companies in this office and we all
>     share
>     >         the same freeswitch server, I'll call then CompanyA,
>     CompanyB
>     >         and CompanyC. CompanyA, B and C all employ their own
>     >         receptionists, and they are all assigned extension 101 on in
>     >         their respective domains. However, the receptionists all
>     work
>     >         very tightly together and basically share work between them.
>     >         So, I need to assign the BLF status of 101 at CompanyA,
>     >         101 at CompanyB and 101 at CompanyC to indicator lights across all
>     >         phones in the building. My problem is that users in CompanyA
>     >         can't see the status of 101 at CompanyB or 101 at CompanyC.
>     Users in
>     >         CompanyB can't see 101 at CompanyA or 101 at CompanyC and so on...
>     >
>     >
>     >
>     >         So, my question is, how do I allow users of one domain in a
>     >         multi-tenant configuration to see the status of other
>     users in
>     >         a different domain?
>     >
>     >
>     >
>     >        
>     _________________________________________________________________________
>     >         Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     >
>     >         
>     > 
>     >
>     >         Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     >         FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >        
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     >    
>     _________________________________________________________________________
>     >     Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     >
>     >     
>     > 
>     >
>     >     Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     >     FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     >
>     >    
>     ------------------------------------------------------------------------
>     >
>     >     If this email is spam, report it to www.OnlyMyEmail.com
>     <http://www.OnlyMyEmail.com>
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>     <https://support.onlymyemail.com/view/report_spam/MTM3Mjc1OjE0MTI0MDQ4MjI6cmlja0BjbGFyaXR5Y29tcHV0ZXIuY29tOmRlbGl2ZXJlZA>
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>     >
>     >
>     >    
>     _________________________________________________________________________
>     >     Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     >
>     >     
>     > 
>     >
>     >     Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     >     FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > <mailto:FreeSWITCH-users at lists.freeswitch.org
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>     > http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Brian D. Foster
>     > Endigo Computer LLC
>     > Email: bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>
>     <mailto:bdfoster at endigotech.com <mailto:bdfoster at endigotech.com>>
>     > Phone: 317-800-7876 <tel:317-800-7876>
>     > Indianapolis, Indiana, USA
>     >
>     > This message contains confidential information and is intended for
>     > those listed in the "To:", "CC:", and/or "BCC:" fields of the
>     message
>     > header. If you are not the intended recipient you are notified that
>     > disclosing, copying, distributing or taking any action in
>     reliance on
>     > the contents of this information is strictly prohibited. E-mail
>     > transmission cannot be guaranteed to be secure or error-free as
>     > information could be intercepted, corrupted, lost, destroyed, arrive
>     > late or incomplete, or contain viruses. The sender therefore
>     does not
>     > accept liability for any errors or omissions in the contents of this
>     > message, which arise as a result of e-mail transmission. If
>     > verification is required please request a hard-copy version.
>     >
>     >
>     ------------------------------------------------------------------------
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From bhegades at gmail.com  Sun Apr 29 21:56:18 2012
From: bhegades at gmail.com (Mahendra Bhegade)
Date: Sun, 29 Apr 2012 10:56:18 -0700
Subject: [Freeswitch-users] Help - SIP gateway issue with fs_cli but not
	with freeswtich console.
Message-ID: <CAC7mRF=jgLBq1ACt5V+tAUUBSmXTApuYMQq55MFcwgT97odH+w@mail.gmail.com>

Hi ,

I was wondering if you can help me with the issue that I am facing. I have
a lua script which dials out using a gateway and works fine from freeswitch
console. If I execute the same script from fs_cli I get an error
identifiying an error with gateway and the complains about the numbe being
invalued

I am using a flowroute gateway and sip end point is
sofia/gateway/flowroute/15104496650 my entry for flowroute.xml is under
conf/sip_profiles/external

 If some one can shed some light on the matter I would appreciate it.



Mahendra

510-449-6650
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From anthony.minessale at gmail.com  Mon Apr 30 02:29:07 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 29 Apr 2012 17:29:07 -0500
Subject: [Freeswitch-users] rfc2833 first Marker Bit not set to True on
	bleg
In-Reply-To: <CAOte9J5PvrAazPTP00Pf6LLNh+uLgLxxA7qsncN-AQUC2_xYhQ@mail.gmail.com>
References: <CAOte9J4kRY50p+Hj7+gxKivvtqYrU13S+UNPhydAh5APy0d45w@mail.gmail.com>
	<CAKV-jFzFvCCjCM6_z7np5FYUc1_Hr4Q__=ga05w5+VXKngsERA@mail.gmail.com>
	<2825583103416496097@unknownmsgid>
	<CAOte9J5PvrAazPTP00Pf6LLNh+uLgLxxA7qsncN-AQUC2_xYhQ@mail.gmail.com>
Message-ID: <CAKbxfG-tvxJsMTXnn3cAy9gFrt9PHRHmP09WmG2ip3fFGj9woA@mail.gmail.com>

Ironicly, it is a bug but a bug we emulate on purpose because Cisco for 6
years has required that incorrect behaviour to work right.  Maybe its
finally time to pull it from the defaults... =p
On Apr 29, 2012 5:19 AM, "Bernard Murphy" <bernard.david.murphy at gmail.com>
wrote:

> Thank-you, <param name="auto-rtp-bugs" value="clear"/> does indeed fix the
> problem of Bleg, first bit Marker not set to True for a Cisco 2901 SIP
> Gateway IOS 15.0(1r)M12
>
> I appreciate the assistance from the group as ever.
>
> Regards
> Bernie
>
>
> On Sun, Apr 29, 2012 at 4:36 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Not a bug!  Set rtp bugs param to clear!
>>
>> /b
>>
>> Sent from my iPhone
>>
>> On Apr 28, 2012, at 4:19 PM, Brian Foster <bdfoster at endigotech.com>
>> wrote:
>>
>> Please report bugs to JIRA. The mailing list is not the place for
>> reporting bugs.
>>
>> Thread closed.
>>
>> -BDF
>> On Apr 28, 2012 5:54 AM, "Bernard Murphy" <bernard.david.murphy at gmail.com>
>> wrote:
>>
>>> On the very latest GIT and previous , Freeswitch doesn't seem to set the
>>> first Marker Bit to true on an rfc2833 event on the bleg of a call out to a
>>> gateway
>>>
>>> on the A-leg the first event from my UA to Freeswitch is correct with
>>> duration as 0 and Marker bit set to True. Then the second event increments
>>> the duration and Market bit is set to false.
>>> but on the B-leg from freeswitch to the cisco gateway the first event
>>> has duration 80 (may or may not be a problem) but the Marker bit is set to
>>> False which must be wrong ? Duration then increments and Marker bit remains
>>> false.
>>>
>>> I also tried <param name="manual-rtp-bugs" value="START_SEQ_AT_ZERO"/>
>>> to make the duration start at zero but it seemed to have no effect.
>>>
>>>
>>> *ALEG from SIP UA to FREESWITCH*
>>> No.     Time Source                Destination           Protocol Length
>>> Info
>>>    5254 23   10.0.47.110           10.50.47.30           RTP EVENT
>>> 60     Payload type=RTP Event, DTMF One 1
>>>
>>> Frame 5254: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
>>> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
>>> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
>>> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
>>> 10.50.47.30 (10.50.47.30)
>>> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
>>> Real-Time Transport Protocol
>>>     [Stream setup by SDP (frame 1122)]
>>>         [Setup frame: 1122]
>>>         [Setup Method: SDP]
>>>     10.. .... = Version: RFC 1889 Version (2)
>>>     ..0. .... = Padding: False
>>>     ...0 .... = Extension: False
>>>     .... 0000 = Contributing source identifiers count: 0
>>>     1... .... = Marker: True
>>>     Payload type: telephone-event (101)
>>>     Sequence number: 26364
>>>     [Extended sequence number: 91900]
>>>     Timestamp: 133920
>>>     Synchronization Source identifier: 0x03bf0676 (62850678)
>>> RFC 2833 RTP Event
>>>     Event ID: DTMF One 1 (1)
>>>     0... .... = End of Event: False
>>>     .0.. .... = Reserved: False
>>>     ..00 1010 = Volume: 10
>>>     Event Duration: 0
>>>
>>> *ALEG SECOND EVENT FROM SIP UA TO FREESWITCH*
>>>
>>> No.     Time Source                Destination           Protocol Length
>>> Info
>>>    5255 23   10.0.47.110           10.50.47.30           RTP EVENT
>>> 60     Payload type=RTP Event, DTMF One 1
>>>
>>> Frame 5255: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
>>> Ethernet II, Src: Sonicwal_3d:2b:1c (00:06:b1:3d:2b:1c), Dst:
>>> Vmware_ab:60:ec (00:0c:29:ab:60:ec)
>>> Internet Protocol Version 4, Src: 10.0.47.110 (10.0.47.110), Dst:
>>> 10.50.47.30 (10.50.47.30)
>>> User Datagram Protocol, Src Port: 49194 (49194), Dst Port: 21374 (21374)
>>> Real-Time Transport Protocol
>>>     [Stream setup by SDP (frame 1122)]
>>>         [Setup frame: 1122]
>>>         [Setup Method: SDP]
>>>     10.. .... = Version: RFC 1889 Version (2)
>>>     ..0. .... = Padding: False
>>>     ...0 .... = Extension: False
>>>     .... 0000 = Contributing source identifiers count: 0
>>>     0... .... = Marker: False
>>>     Payload type: telephone-event (101)
>>>     Sequence number: 26365
>>>     [Extended sequence number: 91901]
>>>     Timestamp: 133920
>>>     Synchronization Source identifier: 0x03bf0676 (62850678)
>>> RFC 2833 RTP Event
>>>     Event ID: DTMF One 1 (1)
>>>     0... .... = End of Event: False
>>>     .0.. .... = Reserved: False
>>>     ..00 1010 = Volume: 10
>>>     Event Duration: 160
>>>
>>> *BLEG FIRST EVENT FROM FREESWITCH* *TO CISCO*
>>> No.     Time Source                Destination           Protocol Length
>>> Info
>>>    5329 23   10.50.47.30           10.50.47.19           RTP EVENT
>>> 58     Payload type=RTP Event, DTMF One 1
>>>
>>> Frame 5329: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
>>> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
>>> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
>>> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
>>> 10.50.47.19 (10.50.47.19)
>>> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
>>> Real-Time Transport Protocol
>>>     [Stream setup by SDP (frame 1113)]
>>>         [Setup frame: 1113]
>>>         [Setup Method: SDP]
>>>     10.. .... = Version: RFC 1889 Version (2)
>>>     ..0. .... = Padding: False
>>>     ...0 .... = Extension: False
>>>     .... 0000 = Contributing source identifiers count: 0
>>>     0... .... = Marker: False
>>>     Payload type: telephone-event (101)
>>>     Sequence number: 24716
>>>     [Extended sequence number: 90252]
>>>     Timestamp: 158240
>>>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
>>> RFC 2833 RTP Event
>>>     Event ID: DTMF One 1 (1)
>>>     0... .... = End of Event: False
>>>     .0.. .... = Reserved: False
>>>     ..00 0111 = Volume: 7
>>>     Event Duration: 80
>>>
>>> *BLEG SECOND EVENT FROM FREESWITCH* *TO CISCO*
>>> No.     Time Source                Destination           Protocol Length
>>> Info
>>>    5331 23   10.50.47.30           10.50.47.19           RTP EVENT
>>> 58     Payload type=RTP Event, DTMF One 1
>>>
>>> Frame 5331: 58 bytes on wire (464 bits), 58 bytes captured (464 bits)
>>> Ethernet II, Src: Vmware_ab:60:ec (00:0c:29:ab:60:ec), Dst:
>>> 64:9e:f3:11:ff:60 (64:9e:f3:11:ff:60)
>>> Internet Protocol Version 4, Src: 10.50.47.30 (10.50.47.30), Dst:
>>> 10.50.47.19 (10.50.47.19)
>>> User Datagram Protocol, Src Port: 24034 (24034), Dst Port: 18768 (18768)
>>> Real-Time Transport Protocol
>>>     [Stream setup by SDP (frame 1113)]
>>>         [Setup frame: 1113]
>>>         [Setup Method: SDP]
>>>     10.. .... = Version: RFC 1889 Version (2)
>>>     ..0. .... = Padding: False
>>>     ...0 .... = Extension: False
>>>     .... 0000 = Contributing source identifiers count: 0
>>>     0... .... = Marker: False
>>>     Payload type: telephone-event (101)
>>>     Sequence number: 24717
>>>     [Extended sequence number: 90253]
>>>     Timestamp: 158240
>>>     Synchronization Source identifier: 0x57e5d7a1 (1474680737)
>>> RFC 2833 RTP Event
>>>     Event ID: DTMF One 1 (1)
>>>     0... .... = End of Event: False
>>>     .0.. .... = Reserved: False
>>>     ..00 0111 = Volume: 7
>>>     Event Duration: 160
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From anthony.minessale at gmail.com  Mon Apr 30 02:36:24 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 29 Apr 2012 17:36:24 -0500
Subject: [Freeswitch-users] Help - SIP gateway issue with fs_cli but not
 with freeswtich console.
In-Reply-To: <CAC7mRF=jgLBq1ACt5V+tAUUBSmXTApuYMQq55MFcwgT97odH+w@mail.gmail.com>
References: <CAC7mRF=jgLBq1ACt5V+tAUUBSmXTApuYMQq55MFcwgT97odH+w@mail.gmail.com>
Message-ID: <CAKbxfG-F_xM3NfntnmVtS7cyma0vKvdFrrAKxh2VMOpeu=Beug@mail.gmail.com>

You would need to supply some details like issue sofia global siptrace on
and console loglevel debug and capture the logs
On Apr 29, 2012 4:30 PM, "Mahendra Bhegade" <bhegades at gmail.com> wrote:

> Hi ,
>
> I was wondering if you can help me with the issue that I am facing. I have
> a lua script which dials out using a gateway and works fine from freeswitch
> console. If I execute the same script from fs_cli I get an error
> identifiying an error with gateway and the complains about the numbe being
> invalued
>
> I am using a flowroute gateway and sip end point is
> sofia/gateway/flowroute/15104496650 my entry for flowroute.xml is under
> conf/sip_profiles/external
>
>  If some one can shed some light on the matter I would appreciate it.
>
>
>
> Mahendra
>
> 510-449-6650
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From msc at freeswitch.org  Mon Apr 30 07:15:48 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 29 Apr 2012 20:15:48 -0700
Subject: [Freeswitch-users] Time routing issue
In-Reply-To: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
References: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
Message-ID: <CAKzWOxXabQUXMbSqPtXii3B345yHdqUYfTY-ksm6bP9LJ-RLUw@mail.gmail.com>

On Fri, Apr 27, 2012 at 9:25 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
>
> I'd like to set a channel variable and use it in a time condition like
> this:
>
> <condition mon="${rmon}" >
>
> with rmon=1-12 (just to be sure condition is verified).
>
> I can't find out why it doesn't work.
>
> - Using <action application="info"/> , I can see variable is set [1-12]
> - I've tried also <condition mon="${rmon}"  inline="true">
> - Using directly  <condition mon="1-12" > it works
>
>
Like Jeff says, be sure that the ${rmon} variable is getting set with the
inline="true" attribute. Note: You don't use inline="true" in a condition
tag.

Try this:
<condition field="${rmon}" expression="^[12]?[0-9]$">

You need to use a regular expression when checking the value of a channel
variable and you need to wrap the the chan var in ${}. Check this section
of the XML Dialplan wiki page:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Caller_Profile_Fields_vs._Channel_Variables

Let us know how it goes!
-MC
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From msc at freeswitch.org  Mon Apr 30 07:15:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 29 Apr 2012 20:15:21 -0700
Subject: [Freeswitch-users] Time routing issue
In-Reply-To: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
References: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
Message-ID: <CAKzWOxVNE-R8obCr1xNpcU-0H+UNde2UgbBPo4Qh8FLNA7z80g@mail.gmail.com>

On Fri, Apr 27, 2012 at 9:25 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
>
> I'd like to set a channel variable and use it in a time condition like
> this:
>
> <condition mon="${rmon}" >
>
> with rmon=1-12 (just to be sure condition is verified).
>
> I can't find out why it doesn't work.
>
> - Using <action application="info"/> , I can see variable is set [1-12]
> - I've tried also <condition mon="${rmon}"  inline="true">
> - Using directly  <condition mon="1-12" > it works
>
>
Like Jeff says, be sure that the ${rmon} variable is getting set with the
inline="true" attribute. Note: You don't use inline="true" in a condition
tag.

Try this:
<condition field="${rmon}" expression="^[12]?[0-9]$">

You need to use a regular expression when checking the value of a channel
variable and you need to wrap the the chan var in ${}. Check this section
of the XML Dialplan wiki page:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Caller_Profile_Fields_vs._Channel_Variables

Let us know how it goes!
-MC
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From msc at freeswitch.org  Mon Apr 30 07:15:21 2012
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 29 Apr 2012 20:15:21 -0700
Subject: [Freeswitch-users] Time routing issue
In-Reply-To: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
References: <1335543926.59940.YahooMailNeo@web132302.mail.ird.yahoo.com>
Message-ID: <CAKzWOxVNE-R8obCr1xNpcU-0H+UNde2UgbBPo4Qh8FLNA7z80g@mail.gmail.com>

On Fri, Apr 27, 2012 at 9:25 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
>
> I'd like to set a channel variable and use it in a time condition like
> this:
>
> <condition mon="${rmon}" >
>
> with rmon=1-12 (just to be sure condition is verified).
>
> I can't find out why it doesn't work.
>
> - Using <action application="info"/> , I can see variable is set [1-12]
> - I've tried also <condition mon="${rmon}"  inline="true">
> - Using directly  <condition mon="1-12" > it works
>
>
Like Jeff says, be sure that the ${rmon} variable is getting set with the
inline="true" attribute. Note: You don't use inline="true" in a condition
tag.

Try this:
<condition field="${rmon}" expression="^[12]?[0-9]$">

You need to use a regular expression when checking the value of a channel
variable and you need to wrap the the chan var in ${}. Check this section
of the XML Dialplan wiki page:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Caller_Profile_Fields_vs._Channel_Variables

Let us know how it goes!
-MC
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From sharad at coraltele.com  Mon Apr 30 08:39:10 2012
From: sharad at coraltele.com (Sharad Garg)
Date: Mon, 30 Apr 2012 10:09:10 +0530
Subject: [Freeswitch-users] wrong detection dtmf rfc2833
References: <4F9AADB5.9060200@gmail.com>
Message-ID: <4B7A607EFC344FA5B305611131BA1647@sharad>

Hi

Where in your dialplan, you enable the DTMF detection, before that, just 
disable once & than enable.

Try doing by this...this may resolve the issue.

Regards
Sharad

 ----- Original Message ----- 
From: "Lubimov Alexey" <avlubimov at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, April 27, 2012 8:01 PM
Subject: [Freeswitch-users] wrong detection dtmf rfc2833


> Good day!
>
> I have problem with dtmf on incoming calls.
>
> Internal dtmf work normal, but on incoming call i have many mistakes
> (111  or 121 instead 110, for example).
>
> siptrace wrong detection attached, Abonent send 110, but freeswitch
> detect 111.
>
> in profile's config i set these options. (Im trying many variants, but
> no find solution)
>
> <param name="dtmf-duration" value="2000"/>
> <param name="dtmf-type" value="rfc2833"/>
> <param name="rfc2833-pt" value="0"/>
> <param name="liberal-dtmf" value="true"/>
>
> Have you any ideas?
>
>
>
> 



From chrisbware at yahoo.it  Mon Apr 30 11:50:26 2012
From: chrisbware at yahoo.it (Chris B. Ware)
Date: Mon, 30 Apr 2012 08:50:26 +0100 (BST)
Subject: [Freeswitch-users] Time routing issue
Message-ID: <1335772226.89474.YahooMailNeo@web132306.mail.ird.yahoo.com>

Hi,

thanks Jeff and Michael for your help.
Of course I set my variable with inline="true". To explain better, I have a catch-all extensions that call a lua script to initialize a set of chan variables according to the called number (destination_number). The script is executed inline.

After that I put a lot of application=log on my dialplan and I can see ${rmon} correctly set. One of my log apps is the anti-action on the time routing condition, so to check ${rmon} when it fails. And has the right value!

I'm wondering if mon,wday,time-of-day can be match against a variable.

I think I have to use reg-exp or lua ?to do it :(?
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From awais.nazeer at hotmail.com  Mon Apr 30 14:23:38 2012
From: awais.nazeer at hotmail.com (awais nazir)
Date: Mon, 30 Apr 2012 15:23:38 +0500
Subject: [Freeswitch-users] unable to load module. mod_gsmopen.so ( Invalid
	ELF header )
Message-ID: <BLU169-W5058857D30A27DE9573E528B280@phx.gbl>





hi, i am unable to load module, when i try to load modules (mod_gsmopen) in freeswitch CLI. seem error like ....   freeswitc->freeswitc->freeswitc-> load mod_gsmopen+OK Reloading XML
-ERR [module load file routine returned an error]2012-04-30 15:34:14.843048 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so
**/usr/local/freeswitch/mod/mod_gsmopen.so: invalid ELF header**
2012-04-30 15:34:14.843048 [INFO] mod_enum.c:812 ENUM Reloaded
 freeswitc-> freeswitc-> !Please help 		 	   		  
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From gmaruzz at gmail.com  Mon Apr 30 15:19:03 2012
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 30 Apr 2012 13:19:03 +0200
Subject: [Freeswitch-users] unable to load module. mod_gsmopen.so (
 Invalid ELF header )
In-Reply-To: <BLU169-W5058857D30A27DE9573E528B280@phx.gbl>
References: <BLU169-W5058857D30A27DE9573E528B280@phx.gbl>
Message-ID: <CALXCt0r=nG+NeYPEzTCMKh5eZtL=FP=eCjmHhak-TcdHqPfWeA@mail.gmail.com>

seems you had problems in compiling the module, the module was not
compiled correctly.

Do the building again, following closely, step by step, the
instruction here: http://wiki.freeswitch.org/wiki/Gsmopen#Building

On Mon, Apr 30, 2012 at 12:23 PM, awais nazir <awais.nazeer at hotmail.com> wrote:
> hi,
>
> i am unable to load module, when i try to load modules?(mod_gsmopen) in
> freeswitch CLI. seem error like ....
>
>
>
> freeswitc->
> freeswitc->
> freeswitc-> load mod_gsmopen
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 2012-04-30 15:34:14.843048 [CRIT] switch_loadable_module.c:1300 Error
> Loading module /usr/local/freeswitch/mod/mod_gsmopen.so
> **/usr/local/freeswitch/mod/mod_gsmopen.so: invalid ELF header**
> 2012-04-30 15:34:14.843048 [INFO] mod_enum.c:812 ENUM Reloaded
>
>
> freeswitc->
>
> freeswitc->
>
> !
> Please help
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From william.suffill at gmail.com  Mon Apr 30 18:53:38 2012
From: william.suffill at gmail.com (William Suffill)
Date: Mon, 30 Apr 2012 10:53:38 -0400
Subject: [Freeswitch-users] Eaves Drop
In-Reply-To: <CAKbxfG9oh_m8VfFgeZOpGnWJGJ-GZ4B=iE0ww2xr4N3mEb2nAg@mail.gmail.com>
References: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local>
	<CAKbxfG9oh_m8VfFgeZOpGnWJGJ-GZ4B=iE0ww2xr4N3mEb2nAg@mail.gmail.com>
Message-ID: <CAF-ZooTWyJtZq8N6v-dA1nzACb70ZMyRWvVdPtBMKD+X697MNA@mail.gmail.com>

uuid_buglist
List the media bugs on channel
Usage: uuid_buglist <uuid>

from mod_commands

dptools has remove_bugs command for the current channel.


From msc at freeswitch.org  Mon Apr 30 19:32:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 30 Apr 2012 08:32:44 -0700
Subject: [Freeswitch-users] Freeswitch speaks italian
In-Reply-To: <1335262363.89683.YahooMailNeo@web132305.mail.ird.yahoo.com>
References: <1335262363.89683.YahooMailNeo@web132305.mail.ird.yahoo.com>
Message-ID: <CAKzWOxVJMoOALF=mcVF5SiOrgnfOLyoLOWADC5mGUXDcA8djqw@mail.gmail.com>

Excellent work, thanks. BTW, can you update the wiki to include this
information? I'd recommend updating this page with information on how to
download the Italian prompts and how to set the language properly for the
say application:

http://wiki.freeswitch.org/wiki/Speech_Phrase_Management

Thanks,
MC

On Tue, Apr 24, 2012 at 3:12 AM, Chris B. Ware <chrisbware at yahoo.it> wrote:

> Hi,
>
> some time ago, Mr. Marco Menardi and his wife Paola (the voice) made a set
> of audio files for Asterisk in italian, under creative commons license.
> Using Audacity I rearranged them to be used with FS.  Translated folders
> are:
>
> ascii
> currency
> digits
> phonetic-ascii
> time
> voicemail
>
> Only 8Khz version, sorry.  Link to my blog, where you can download files:
> http://chrisbware.wordpress.com/2012/04/12/freeswitch-parla-italiano/
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From mario_fs at mgtech.com  Mon Apr 30 20:11:53 2012
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 30 Apr 2012 09:11:53 -0700
Subject: [Freeswitch-users] Forward incoming call to cell phone if user
	not reachable/registered
In-Reply-To: <CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>
References: <CAJ_WQfF8k3tSa602YMCPukj1o3POv_e1dBz8e36P_ksa0Roa8A@mail.gmail.com>
	<4F9B1CDE.6040509@quentustech.com>
	<CAKV-jFxyMpGn7XGV4uEk7GR-1mjcwPuFzS_7+LrPZr7Nz2Q6Nw@mail.gmail.com>
	<CAJ_WQfFk9sPdkL6uBy0==2yo9M0mf7cT6++3OSvoPkUrcQ=HdA@mail.gmail.com>
Message-ID: <B73AB2B5-74E6-471C-B674-2F32BBEF98C0@mgtech.com>

I have something similar, in my case if a caller comes in I ring the inside extensions for 17 seconds, if no answer I ring for 20 more seconds and 2 cell phones at the same time, finally, if no answer go to voicemail. The diaplan has continue-on-fail set to true so if the first 17 second bridge fails it goes to the second bridge, and if that fails voicemail.

The difference from you is that you don't want to call the cell phone UNLESS the local extension if not registered. I would try this: put the call into one bridge command and use the "|" character between the local extension and cell phone which causes a sequential call. That way if the local extension is not available the call to it will fail and roll over to the cell phone. The snippet below is illustrates the important stuff. The difference from you is that I am calling all phones and the cell phones at the same time in the second bridge using the ":_:" (enterprise) syntax. If they are replaced by "|" the calls would be sequential (you also need to change <originate_timeout..> since <> only applies to enterprise syntax). Hope this helps a little.

I also found this... http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_8:_Check_user_is_authenticated 
Mario G

        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="{originate_timeout=17}${group_call(everyone@${domain_name}+A)}"/>
        <!-- this is needed to allow call_timeout to work after bridging to a gateway -->
        <action application="set" data="ignore_early_media=true"/>
        <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>${group_call(everyone@${domain_name}+E)}:_:sofia/gateway/${dial_gateway}/19161111111:_:sofia/gateway/${dial_gateway}/19162222222"/>
        <!-- Timeout/No answer go to voicemail -->
        <action application="playback" data="file_string://${xxsounds}please_leave_a_message.wav!//${sayname}"/>
        <action application="answer"/>
        <!-- Next line removes default greeting, we already played one above-->
        <action application="set" data="skip_greeting=true"/>
        <action application="voicemail" data="default ${domain_name} 100"/>


On Apr 28, 2012, at 6:46 AM, Travis Langhals wrote:

> Thanks for the info, I have the voicemail action working but if I add a second bridge command between my extension bridge and voicemail actions it rings the cell phone if the extension doesn't answer within my 20s timeout.  I'm trying to make it only ring the cell phone if the extension is not online and then go to voicemail.  If the extension is online it should just ring the extension and then go to voicemail.
> 
> On Fri, Apr 27, 2012 at 9:39 PM, Brian Foster <bdfoster at endigotech.com> wrote:
> Look at the local exensions dialplan in default.xml for an example, specifically how it is forwarded on to voicemail. You could get slick and do conditions and set a param inside of the users .xml file in the directory to tell the dialplan where to try next. Also look at the call_timeout variable.
> 
> -BDF
> 
> On Apr 27, 2012 6:26 PM, "William King" <william.king at quentustech.com> wrote:
> You want to do research into the continue-on-fail  and the variable hangup after bridge. 
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> 
> Basically you can have multiple bridge actions in an extension and the first to successfully bridge a call will be the last action the call will make.
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
> 
> On 04/27/2012 03:01 PM, Travis Langhals wrote:
>> 
>> Could someone please help with an example dialplan for call failover to an external DID (i.e. cell phone or pots line) in case the the user has a network failure can cannot be reached?
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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From marketing at cluecon.com  Mon Apr 30 20:48:04 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 30 Apr 2012 09:48:04 -0700
Subject: [Freeswitch-users] ClueCon 2012 Preview: Vestec Automatic Speech
	Recognition
Message-ID: <CAKzWOxWFmeB=koYeT=XMV-15VpzczBMS0N44c34vDSs3tCPQEA@mail.gmail.com>

Greetings!

This week we have a small taste of ClueCon 2012: The Vestec
<http://www.vestec.com?cc120430>team will be joining us on the
FreeSWITCH community
conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call
this Wednesday, May 2. Vestec produces high-quality automatic speech
recognition (ASR) software for open source telephony projects like Asterisk
and FreeSWITCH. We look forward to having a Vestec engineer on the call
with us this week to give us a technical look at their product and to
answer community questions. Vestec will also be at ClueCon 2012 as a Silver
Sponsor and have committed to being available for one-on-one consulting for
those wanting to discuss their specific projects. Yet another reason to
join us at ClueCon this August!

The ClueCon 2012 speakers list is starting to take shape. We have a number
of "fan-favorites" schedule as well as a few new surprises. We are pleased
to have returning for a third consecutive year Mr. Philip Zimmermann,
inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the
ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously
announced, ZRTP is now included in the FreeSWITCH distribution. Philip will
be discussing VoIP security in general, as well developments with the ZRTP
project.

Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today
and take advantage of ClueCon Power Play, which gives you extra entries in
the world famous giveaways we have each year. Feel free to contact us if
you have any questions about attending ClueCon, sponsoring, or speaking at
the event.

We look forward to seeing you there - in less than 100 days!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12-0430
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From marketing at cluecon.com  Mon Apr 30 20:59:48 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 30 Apr 2012 09:59:48 -0700
Subject: [Freeswitch-users] ClueCon 2012 Preview: Vestec Automatic Speech
	Recognition
Message-ID: <CAKzWOxUUZt6krVjtuJp3u4urA8xc7UZo76y_7wqD7OCR6xVuWg@mail.gmail.com>

Greetings!

This week we have a small taste of ClueCon 2012: The Vestec
<http://www.vestec.com?cc120430>team will be joining us on the
FreeSWITCH community
conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call
this Wednesday, May 2. Vestec produces high-quality automatic speech
recognition (ASR) software for open source telephony projects like Asterisk
and FreeSWITCH. We look forward to having a Vestec engineer on the call
with us this week to give us a technical look at their product and to
answer community questions. Vestec will also be at ClueCon 2012 as a Silver
Sponsor and have committed to being available for one-on-one consulting for
those wanting to discuss their specific projects. Yet another reason to
join us at ClueCon this August!

The ClueCon 2012 speakers list is starting to take shape. We have a number
of "fan-favorites" scheduled as well as a few new surprises. We are pleased
to have returning for a third consecutive year Mr. Philip Zimmermann,
inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the
ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously
announced <http://freeswitch.org/node/388?cc12-0430>, ZRTP is now included
in the FreeSWITCH distribution. Philip will be discussing VoIP security in
general, as well developments with the ZRTP project.

Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today
and take advantage of ClueCon Power Play, which gives you extra entries in
the world famous giveaways we have each year. Feel free to contact us if
you have any questions about attending ClueCon, sponsoring, or speaking at
the event.

We look forward to seeing you there - in less than 100 days!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
cc12-0430
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From kerem.erciyes at gmail.com  Mon Apr 30 22:52:08 2012
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Mon, 30 Apr 2012 21:52:08 +0300
Subject: [Freeswitch-users] How to update a live system with very little
 downtime (How does 'make current' work?)
In-Reply-To: <CAKzWOxUaAv3zSkmej1ansZMK2k_hAcM+J-OLy_8t=6Wj51hgkw@mail.gmail.com>
References: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local>
	<CAKzWOxUaAv3zSkmej1ansZMK2k_hAcM+J-OLy_8t=6Wj51hgkw@mail.gmail.com>
Message-ID: <CAJ6wjQEaTB4QM4OTHY-R36q1vAtWerxfyQe_qyG_3jLOZ0=ehA@mail.gmail.com>

Or you could always build a debian-live image that includes
freeswitch-current and all the wanpipe driver etc, and swap them out at
will at 5 mins downtime maximum, store all you data on HDD of /etc and
/freeswitch/conf and you are good to go....

On Tue, Apr 17, 2012 at 12:02 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Mon, Apr 16, 2012 at 12:01 PM, Jason Moran <jmoran at secureachsystems.com
> > wrote:
>
>>  Sorry for such a basic question:****
>>
>> **1.       **If I do a ?make current? while a system is running it will
>> not have any effect until I restart FS, correct?****
>>
>> **2.       **What if ?make current? fails? What if it fails and the
>> server gets restarted, will FS work?****
>>
>> **3.       **Is it more advisable to stop FS, do the ?make current?,
>> then start FS even though it will have a longer down time?****
>>
>> ** **
>>
>> I know it?s best to keep FS up-to-date ? but it is difficult when you
>> only have 1 system available and it is expected to have 100% uptime..****
>>
>> Thanks,
>> Jason
>>
>
> Since you don't have a spare system (which is totally naughty, btw,
> especially if your corporate overlords expect 100% uptime) then my next
> question would be this: how are you set on CPU power, RAM, and disk space?
> The reason I ask is that it is entirely possible to create a secondary git
> clone of the FS repo and use a completely different install path when
> running the configure script. You could "make current" in that secondary
> repo and get a sort of preview of what would happen w/o touching your
> production build env or install dirs. If the update goes well in your test
> scenario then you could do the update in your production environment.
>
>  It's kinda hackish but if you have spare disk space then it's a really
> cheap option. It isn't perfect, but it does give you at least some
> semblance of "try before you buy."
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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