From codecomplete at free.fr  Sun Mar  1 10:20:54 2009
From: codecomplete at free.fr (Fred)
Date: Sun, 01 Mar 2009 19:20:54 +0100
Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?
Message-ID: <7.0.1.0.2.20090301191907.027bd870@fredshack.com>

Hello

As an easy way to show a Freeswitch server to prospects, I'm thinking 
of buying an Asus notebook along with a Sangom USB FXO gateway.

www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port

If someone's been using those two thingies, I'm curious to know if 
they happily run Freeswitch, or if I should look for some other hardware?

Thank you.



From rex.alex345 at yahoo.com  Sun Mar  1 11:13:34 2009
From: rex.alex345 at yahoo.com (Rex_Alex)
Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST)
Subject: [Freeswitch-users]  To do telephony functions from web page
Message-ID: <1235934814358-2405620.post@n2.nabble.com>


Hi All,

I am trying to do the telephonic functions(like dial, hangup, conference and
etc.) from a webpage (for a customization) rather than doing it from a soft
phone.

What would be the optimal way of doing it?

Please suggest.

Thanks,
Rex.
-- 
View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405620.html
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From krice at freeswitch.org  Sun Mar  1 11:17:39 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 01 Mar 2009 13:17:39 -0600
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <1235934814358-2405620.post@n2.nabble.com>
Message-ID: <C5D03B73.4F7F4%krice@freeswitch.org>

Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control
things.... Both methods work well

K



From: Rex_Alex <rex.alex345 at yahoo.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST)
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users]  To do telephony functions from web page

Hi All, I am trying to do the telephonic functions(like dial, hangup,
conference and etc.) from a webpage (for a customization) rather than doing
it from a soft phone. What would be the optimal way of doing it? Please
suggest. Thanks, Rex.

View this message in context: To do telephony functions from web page
<http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405
620.html> 
Sent from the freeswitch-users mailing list archive
<http://n2.nabble.com/freeswitch-users-f2379917.html>  at Nabble.com.


_______________________________________________
Freeswitch-users mailing 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 rex.alex345 at yahoo.com  Sun Mar  1 11:51:18 2009
From: rex.alex345 at yahoo.com (Rex_Alex)
Date: Sun, 1 Mar 2009 11:51:18 -0800 (PST)
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <C5D03B73.4F7F4%krice@freeswitch.org>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
Message-ID: <1235937078290-2405782.post@n2.nabble.com>


Hi,

Learned how to enable mod_xml_rpc but didn't find any samples. 

Please post me a sample to send requests(like dial) and receive
responses(like uuid) from FreeSWITCH using mod_xml_rpc

Please assist.

Thanks,
Rex.



Ken Rice-2 wrote:
> 
> Check out ESL for PHP, Perl etc, or you can use mod_xml_rpc to control
> things.... Both methods work well
> 
> K
> 
> 
> 
> From: Rex_Alex 
> Reply-To: 
> Date: Sun, 1 Mar 2009 11:13:34 -0800 (PST)
> To: 
> Subject: [Freeswitch-users]  To do telephony functions from web page
> 
> Hi All, I am trying to do the telephonic functions(like dial, hangup,
> conference and etc.) from a webpage (for a customization) rather than
> doing
> it from a soft phone. What would be the optimal way of doing it? Please
> suggest. Thanks, Rex.
> 
> View this message in context: To do telephony functions from web page
>  
> Sent from the freeswitch-users mailing list archive
>   at Nabble.com.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 sicfslist at gmail.com  Sun Mar  1 11:57:03 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sun, 1 Mar 2009 13:57:03 -0600
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <1235937078290-2405782.post@n2.nabble.com>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
	<1235937078290-2405782.post@n2.nabble.com>
Message-ID: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>

Rex:

The basis for xml_rpc or mod_event is something along the lines of:

api $command

As an example to originate a call (using xml_rpc or mod_event) you would do:

api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context

What language are you trying to do this in?

SDR
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From rex.alex345 at yahoo.com  Sun Mar  1 12:10:04 2009
From: rex.alex345 at yahoo.com (Rex_Alex)
Date: Sun, 1 Mar 2009 12:10:04 -0800 (PST)
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
	<1235937078290-2405782.post@n2.nabble.com>
	<35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>
Message-ID: <1235938204874-2405845.post@n2.nabble.com>


Hi Shelby Ramsey,

I would like to do the same in php script.

Please post me a sample.

Thanks,
Rex.


Shelby Ramsey wrote:
> 
> Rex:
> 
> The basis for xml_rpc or mod_event is something along the lines of:
> 
> api $command
> 
> As an example to originate a call (using xml_rpc or mod_event) you would
> do:
> 
> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml $context
> 
> What language are you trying to do this in?
> 
> SDR
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From sicfslist at gmail.com  Sun Mar  1 12:19:16 2009
From: sicfslist at gmail.com (Shelby Ramsey)
Date: Sun, 1 Mar 2009 14:19:16 -0600
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <1235938204874-2405845.post@n2.nabble.com>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
	<1235937078290-2405782.post@n2.nabble.com>
	<35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>
	<1235938204874-2405845.post@n2.nabble.com>
Message-ID: <35b355e90903011219p20ee2e5k9e6d553598393394@mail.gmail.com>

Rex,

I've never actually used PHP for this type of thing ... but you might want
to start by looking here:

http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/php/single_command.php?r=12216

or

http://wiki.freeswitch.org/wiki/PHP_Event_Socket

Good luck.  I'm sure some other folks here who use PHP for this type of app
will be able to assist more.

SDR
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From gmaruzz at celliax.org  Sun Mar  1 12:30:25 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 1 Mar 2009 21:30:25 +0100
Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?
In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com>
References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com>
Message-ID: <7b197bef0903011230k16afc8d1hd43dd224570787f@mail.gmail.com>

I, for one, run often FS  on an eeepc900 (one year old).
NEver tested max concurrent SIP calls, but for sure is able to run concurrently:
- one FS
- two SIP calls
- two Skypiax calls
- two linux Skype client instances
- two Skype calls

Also, I often use it to generate 6 or 8 concurrent Skype calls.

So, taking account of how heavy Skype is, it probably is able to run
FS with dozens concurrent SIP calls.
Gm

On 3/1/09, Fred <codecomplete at free.fr> wrote:
> Hello
>
> As an easy way to show a Freeswitch server to prospects, I'm thinking
> of buying an Asus notebook along with a Sangom USB FXO gateway.
>
> www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port
>
> If someone's been using those two thingies, I'm curious to know if
> they happily run Freeswitch, or if I should look for some other hardware?
>
> Thank you.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039


From csorlie at teldio.com  Sun Mar  1 10:01:21 2009
From: csorlie at teldio.com (Cameron Sorlie)
Date: Sun, 01 Mar 2009 13:01:21 -0500
Subject: [Freeswitch-users] Detecting the origin of voice activity using VAD
Message-ID: <49AACD71.5080103@teldio.com>

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From brian at freeswitch.org  Sun Mar  1 13:15:20 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2009 15:15:20 -0600
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <49AACD71.5080103@teldio.com>
References: <49AACD71.5080103@teldio.com>
Message-ID: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>

If i'm not mistaken those events will have a member-id in them so you  
can tell who they belong to.

/b

On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote:

> Using voice activity detection (VAD) in FreeSWITCH, how might I then  
> distinguish which side of a call any given TALK or NOTALK event  
> relates to? I am interested not just that there is activity on the  
> call, but interested also in which party on the call is speaking (or  
> not).
>
> Cam



From Prometheus001 at gmx.net  Sun Mar  1 13:38:20 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 01 Mar 2009 22:38:20 +0100
Subject: [Freeswitch-users] hide caller_id_number,
	transfer and then change caller_id_number
Message-ID: <49AB004C.6090604@gmx.net>

Hello,

I have the following problem while providing callback (mod_eventsocket
is used):
1) I want to call a certain destination number A with a suppressed
caller_id_number (this works fine with some vars in the origination string)
2) The destination number A picks up the phone and enters a target
number B by DTMF
3) freeswitch then forwards the call to target number B by DTMF and I
want to show the number A. I do this with uuid_setvar. The problem is
that it still shows unknown.
This is all with SIP.

uuid_setvar however worked if I did not set the caller_id_number to
unknown. Per default this is then "00000000000" and can then be changed
with uuid_setvar to the number of A.
But if I set caller_id_number to unknown I can no longer change it to A.

Any hint?

Best regards
Peter







From Prometheus001 at gmx.net  Sun Mar  1 14:27:25 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 01 Mar 2009 23:27:25 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>
References: <49A92BAE.4090907@gmx.net>
	<F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>
Message-ID: <49AB0BCD.8030108@gmx.net>

Hello Brian,

thanks for the info. I am a step further, but it cannot load the grammar
files.
I am sending through event_socket:

SendMsg
call-command: execute
execute-app-name: detect_speech
execute-app-arg: pocketsphinx yes no

However I get the message (also when I am using Pizza demo):
2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
sofia/internal/1000 at sip2.server.com Command Execute
detect_speech(pocketsphinx yes no)
2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
pocketsphinx_asr_load_grammar() Can't open language model
/usr/local/freeswitch/grammar/model/communicator.
2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
switch_ivr_detect_speech() Error loading Grammar
2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
pocketsphinx_asr_close() Port Closed.

However the grammar files are there:
root at sip2:/usr/local/freeswitch/grammar/model/communicator#
root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
total 12752
drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
-rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
-rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
-rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
-rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
-rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
-rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
-rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
-rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances


Any hint?

Best regards
Peter

Brian West schrieb:
> You can accomplish this .... here is an example using ESL in perl
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344
>
> /b
>
> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:
>
>   
>> Or back to the basics: Is it possible to use pocketsphinx through  
>> event
>> socket?
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mike at jerris.com  Sun Mar  1 16:34:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 1 Mar 2009 19:34:30 -0500
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <1235938204874-2405845.post@n2.nabble.com>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
	<1235937078290-2405782.post@n2.nabble.com>
	<35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>
	<1235938204874-2405845.post@n2.nabble.com>
Message-ID: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com>

There are examples on the wiki for this.

Mike

On Mar 1, 2009, at 3:10 PM, Rex_Alex <rex.alex345 at yahoo.com> wrote:

>
> Hi Shelby Ramsey,
>
> I would like to do the same in php script.
>
> Please post me a sample.
>
> Thanks,
> Rex.
>
>
> Shelby Ramsey wrote:
>>
>> Rex:
>>
>> The basis for xml_rpc or mod_event is something along the lines of:
>>
>> api $command
>>
>> As an example to originate a call (using xml_rpc or mod_event) you  
>> would
>> do:
>>
>> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml  
>> $context
>>
>> What language are you trying to do this in?
>>
>> SDR
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 red.rain.seven at gmail.com  Sun Mar  1 18:20:32 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 1 Mar 2009 18:20:32 -0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
Message-ID: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>

Does the freeswitch VAD is able to distinguish ring tone from human voice?
The scenario is to originate a call to a IVR system(don't connect the other
leg here yet) and dial DTMF to get to the designated extension number , once
someone picks up and say hello ( detected by VAD) now release to connect the
other leg of the call. The point is to hold the first leg till a real person
picks up.

If it can't be done by VAD, how should I approach this function that I want
to achieve.

Thanks

On Sun, Mar 1, 2009 at 1:15 PM, Brian West <brian at freeswitch.org> wrote:

> If i'm not mistaken those events will have a member-id in them so you
> can tell who they belong to.
>
> /b
>
> On Mar 1, 2009, at 12:01 PM, Cameron Sorlie wrote:
>
> > Using voice activity detection (VAD) in FreeSWITCH, how might I then
> > distinguish which side of a call any given TALK or NOTALK event
> > relates to? I am interested not just that there is activity on the
> > call, but interested also in which party on the call is speaking (or
> > not).
> >
> > Cam
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sun Mar  1 18:28:18 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2009 20:28:18 -0600
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
Message-ID: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>

NO.  You want something that people THINK exists and works well...  
Reliable human/voice detection doesn't exist in ANY form.

/b

On Mar 1, 2009, at 8:20 PM, Henry Huang wrote:

> Does the freeswitch VAD is able to distinguish ring tone from human  
> voice?
> The scenario is to originate a call to a IVR system(don't connect  
> the other leg here yet) and dial DTMF to get to the designated  
> extension number , once someone picks up and say hello ( detected by  
> VAD) now release to connect the other leg of the call. The point is  
> to hold the first leg till a real person picks up.
>
> If it can't be done by VAD, how should I approach this function that  
> I want to achieve.
>
> Thanks



From red.rain.seven at gmail.com  Sun Mar  1 19:05:19 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 1 Mar 2009 19:05:19 -0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
Message-ID: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com>

Well, I knew it would be some future fantasy for now..
If not human detection. I guess will try to use Dialplan Tools wait for
silence to wait till the ring tone is finished ,then connect the other leg.






On Sun, Mar 1, 2009 at 6:28 PM, Brian West <brian at freeswitch.org> wrote:

> NO.  You want something that people THINK exists and works well...
> Reliable human/voice detection doesn't exist in ANY form.
>
> /b
>
> On Mar 1, 2009, at 8:20 PM, Henry Huang wrote:
>
> > Does the freeswitch VAD is able to distinguish ring tone from human
> > voice?
> > The scenario is to originate a call to a IVR system(don't connect
> > the other leg here yet) and dial DTMF to get to the designated
> > extension number , once someone picks up and say hello ( detected by
> > VAD) now release to connect the other leg of the call. The point is
> > to hold the first leg till a real person picks up.
> >
> > If it can't be done by VAD, how should I approach this function that
> > I want to achieve.
> >
> > Thanks
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sun Mar  1 19:11:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 1 Mar 2009 21:11:32 -0600
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com>
Message-ID: <F0858B11-A1E5-492E-B507-F3178CE9F07B@freeswitch.org>

Usually ringing is done in early media... so the best bet would be to  
ignore_early_media=true

/b

On Mar 1, 2009, at 9:05 PM, Henry Huang wrote:

> Well, I knew it would be some future fantasy for now..
> If not human detection. I guess will try to use Dialplan Tools wait  
> for silence to wait till the ring tone is finished ,then connect the  
> other leg.
>

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From mrene_lists at avgs.ca  Sun Mar  1 21:00:44 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 2 Mar 2009 00:00:44 -0500
Subject: [Freeswitch-users] Qt portaudio interface
Message-ID: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca>

Hi all,

Anyone interested in contributing to a Qt interface in order to make a  
decent softphone using FS please reply to this thread.
(also give your availability so we can have a conference call to  
decide stuff)

Thanks,
Math


From kawarod at laposte.net  Sun Mar  1 23:06:10 2009
From: kawarod at laposte.net (rod)
Date: Mon, 02 Mar 2009 11:06:10 +0400
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
Message-ID: <49AB8562.4050806@laposte.net>

Hi All,

I ran some longer tests with FS 1.0.3 acting as an SBC.
The test machine has the following specs:
    - Intel Quad Core Q9550
    - 8GB RAM (far too much from what I saw)

After 3 days running SIPP with 750 simultaneous calls (1500 channels) at 
20cps mean (50cps max) and call duration of 35s, I stopped SIPP.

In the CLI, using status command I got this:

freeswitch at internal> status
UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds, 
607 microseconds
15817560 session(s) since startup
22 session(s) 0/500

But when I use "show channels" or "show calls", I see nothing. So I'm 
wondering where are these 22 sessions ?

FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs.

Successful call -->  5271434                
Failed call ---> 1554      (less than 0.03%)       

regards,
rod.



complete SIPP summary:

------------------------------ Scenario Screen -------- [1-9]: Change 
Screen --
  Call-rate(length)     Port   Total-time  Total-calls  Remote-host
50.0(35000 ms)/1.000s   5060  254259.42 s      5273022  
10.10.10.254:5060(UDP)

  0 new calls during 0.856 s period      7 ms scheduler resolution
  0 calls (limit 750)                    Peak was 750 calls, after 15 s
  0 Running, 34 Paused, 0 Woken up
  15544 out-of-call msg (discarded)   
  1 open sockets                       
  9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate 
(kB/s)
  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout   
Unexpected-Msg
      INVITE ---------->         5273022   0         0                 
         100 <----------         5273022   0                   1554    
         180 <----------         0         0                   0       
         183 <----------         0         0                   0       
         200 <----------  E-RTD1 5271434   0                   0       
         ACK ---------->         5271434   0                           
       Pause [    35.0s]         5271434                       0       
         BYE ---------->         5271434   0         0                 
         200 <----------         5271434   0                   0       

------------------------------ Test Terminated 
--------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change 
Screen --
  Start Time             | 2009-02-27 
09:11:31                                 
  Last Reset Time        | 2009-03-02 
07:49:10                                 
  Current Time           | 2009-03-02 
07:49:11                                 
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:857              | 
70:37:39:429            
  Call Rate              |    0.000 cps              |   20.739 
cps            
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        
0                
  OutGoing call created  |        0                  |  
5273022                
  Total Call created     |                           |  
5273022                
  Current Call           |       34                  
|                         
-------------------------+---------------------------+--------------------------
  Successful call        |        0                  |  
5271434                
  Failed call            |        0                  |     
1554                
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:000              | 
00:00:00:240            
  Call Length            | 38:32:13:386              | 
00:00:36:131            
------------------------------ Test Terminated 
--------------------------------




From red.rain.seven at gmail.com  Sun Mar  1 23:18:02 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 1 Mar 2009 23:18:02 -0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <F0858B11-A1E5-492E-B507-F3178CE9F07B@freeswitch.org>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<59ad9ca10903011905j1348513awc5f83c213c57927e@mail.gmail.com>
	<F0858B11-A1E5-492E-B507-F3178CE9F07B@freeswitch.org>
Message-ID: <59ad9ca10903012318k53852016ied8d982a467577c3@mail.gmail.com>

ignore_early_media=true is not going to do the trick since once the IVR
picks up the call on leg A, the ring tone is stopped and the IVR is going to
play pre-recorded voice menu. And the freeswtich is going to send DTMF to
reach a certain extension number say 101. Then the ring tone is going to
start again while the IVR is going to dial the 101 extension(or even play
moh while dialing). After extension 101 picks up, this is when I want the
"originate" to connect call leg B on some other number.

On Sun, Mar 1, 2009 at 7:11 PM, Brian West <brian at freeswitch.org> wrote:

> Usually ringing is done in early media... so the best bet would be to
> ignore_early_media=true
> /b
>
> On Mar 1, 2009, at 9:05 PM, Henry Huang wrote:
>
> Well, I knew it would be some future fantasy for now..
> If not human detection. I guess will try to use Dialplan Tools wait for
> silence to wait till the ring tone is finished ,then connect the other leg.
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From saigop at gmail.com  Mon Mar  2 01:25:23 2009
From: saigop at gmail.com (Gopalakrishnan A.N)
Date: Mon, 2 Mar 2009 14:55:23 +0530
Subject: [Freeswitch-users] To do telephony functions from web page
In-Reply-To: <2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com>
References: <1235934814358-2405620.post@n2.nabble.com>
	<C5D03B73.4F7F4%krice@freeswitch.org>
	<1235937078290-2405782.post@n2.nabble.com>
	<35b355e90903011157o640c4e3bi422e0a2cbaaf4221@mail.gmail.com>
	<1235938204874-2405845.post@n2.nabble.com>
	<2A727DCA-B690-4A30-97EF-4D16223ECE45@jerris.com>
Message-ID: <2ea4d47e0903020125m4be4a5ffl4c33e6f3325a2919@mail.gmail.com>

Hi Rex,
  Please find the attached file for the PHP script. This script has been
executed in FS 1.0.2. put these two scripts in htdocs directory. access the
http://localhost/sample2.php so that two text box will appear. you can able
to give the extension number and mobile number to dial. Try this :)

On Mon, Mar 2, 2009 at 6:04 AM, Michael Jerris <mike at jerris.com> wrote:

> There are examples on the wiki for this.
>
> Mike
>
> On Mar 1, 2009, at 3:10 PM, Rex_Alex <rex.alex345 at yahoo.com> wrote:
>
> >
> > Hi Shelby Ramsey,
> >
> > I would like to do the same in php script.
> >
> > Please post me a sample.
> >
> > Thanks,
> > Rex.
> >
> >
> > Shelby Ramsey wrote:
> >>
> >> Rex:
> >>
> >> The basis for xml_rpc or mod_event is something along the lines of:
> >>
> >> api $command
> >>
> >> As an example to originate a call (using xml_rpc or mod_event) you
> >> would
> >> do:
> >>
> >> api originate sofia/external/$SOMEANI@$IP:$PORT $EXTENSION xml
> >> $context
> >>
> >> What language are you trying to do this in?
> >>
> >> SDR
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > --
> > View this message in context:
> http://n2.nabble.com/To-do-telephony-functions-from-web-page-tp2405620p2405845.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thank you  with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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From gopal2krishnan at gmail.com  Mon Mar  2 01:29:13 2009
From: gopal2krishnan at gmail.com (Gopal krishnan)
Date: Mon, 2 Mar 2009 14:59:13 +0530
Subject: [Freeswitch-users] ESL Wrapper
In-Reply-To: <1235740392995-2395557.post@n2.nabble.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<49A6E1DB.3070806@freeswitch.org>
	<1235740392995-2395557.post@n2.nabble.com>
Message-ID: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com>

Hi,
  Actually what is the difference between ESL in FS 1.0.3 and event socket
in FS 1.0.2. Is the FS 1.0.3 ESL superior?

On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex <rex.alex345 at yahoo.com> wrote:

> Hi All, I did what you have all suggested. Now its working perfectly.
> Thanks a lot for all your assistance. Rex.
>
> Raymond Chandler wrote:
>  and it will probably be a good idea to do make phpmod-install so that the
> .so and .php files gets into the correct place to be included -Ray Mathieu
> Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at
> 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh &&
> ./configure && make >> install and did Mathieu's suggestion but getting
> error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a"
> >> SOLINK="-shared -Xlinker -x" >>
> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g
> -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror
> -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>
> -Wmissing-prototypes" >>
> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE
> -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]:
> php-config: Command not found make[1]: >> Entering directory
> `/root/freeswitch-1.0.3/libs/esl/php' g++ >>
> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb
> -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >>
> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >>
> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >>
> esl_wrap.cpp:719:17: error: php.h: No such file or directory >>
> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >>
> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >>
> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >>
> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >>
> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >>
> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error:
> expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or
> field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error:
> ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was
> not declared in this scope >> esl_wrap.cpp:793: error: expected
> primary-expression before ?void? >> esl_wrap.cpp:793: error: expected
> primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was
> not declared in this scope >> esl_wrap.cpp:793: error: expected
> primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer
> expression list treated as >> compound expression esl_wrap.cpp:793: error:
> expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make:
> *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i
> wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34
> AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it..
> Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh &&
> ./configure && >> make install And then do Mathieu's suggestion? -MC >>
> _______________________________________________ Freeswitch-users >> mailing
> list Freeswitch-users at lists.freeswitch.org >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>>
> http://www.freeswitch.org >> >> >>
> ------------------------------------------------------------------------ >>
> View this message in context: Re: ESL Wrapper >> >> Sent from the
> freeswitch-users mailing list archive >> at Nabble.com. >>
> _______________________________________________ >> Freeswitch-users mailing
> list >> Freeswitch-users at lists.freeswitch.org >> >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>>
> http://www.freeswitch.org > >
> ------------------------------------------------------------------------ > >
> _______________________________________________ > Freeswitch-users mailing
> list > Freeswitch-users at lists.freeswitch.org >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >
> http://www.freeswitch.org >
> _______________________________________________ Freeswitch-users mailing
> list Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> View this message in context: Re: ESL Wrapper<http://n2.nabble.com/ESL-Wrapper-tp2385651p2395557.html>
> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thank you  with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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From gopal2krishnan at gmail.com  Mon Mar  2 01:42:19 2009
From: gopal2krishnan at gmail.com (Gopal krishnan)
Date: Mon, 2 Mar 2009 15:12:19 +0530
Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?
In-Reply-To: <7.0.1.0.2.20090301191907.027bd870@fredshack.com>
References: <7.0.1.0.2.20090301191907.027bd870@fredshack.com>
Message-ID: <2ea4d47e0903020142n62479f72o996243ea6b5bee64@mail.gmail.com>

Hi Fred,
   Yes you can use Sangoma USB FXO with your laptop. You need to install
openzap for this. But for testing you can use this driver. Still there is
some issue with Openzap with FS as for as I used. while installing Sangoma
USB FXO device you need to use beta drivers.

On Sun, Mar 1, 2009 at 11:50 PM, Fred <codecomplete at free.fr> wrote:

> Hello
>
> As an easy way to show a Freeswitch server to prospects, I'm thinking
> of buying an Asus notebook along with a Sangom USB FXO gateway.
>
> www.telephonydepot.com/Catalog/Sangoma/Sangoma-USB-FXO-U100-2-Port
>
> If someone's been using those two thingies, I'm curious to know if
> they happily run Freeswitch, or if I should look for some other hardware?
>
> Thank you.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thank you  with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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From codecomplete at free.fr  Mon Mar  2 03:47:19 2009
From: codecomplete at free.fr (Fred)
Date: Mon, 02 Mar 2009 12:47:19 +0100
Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?
In-Reply-To: <mailman.6058.1235986943.18013.freeswitch-users@lists.frees
	witch.org>
References: <mailman.6058.1235986943.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <7.0.1.0.2.20090302124612.028657a8@free.fr>

Thanks guys for the feedback. So, the OpenZap driver isn't ready for 
production yet?



From sridhart at alcatel-lucent.com  Mon Mar  2 03:52:10 2009
From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar))
Date: Mon, 2 Mar 2009 17:22:10 +0530
Subject: [Freeswitch-users] Running freeswitch on powerpc
In-Reply-To: <mailman.4507.1233589307.18013.freeswitch-users@lists.freeswitch.org>
References: <mailman.4507.1233589307.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>

Hi all,

I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code

Regards
Sridhar


> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
> Behalf Of freeswitch-users-request at lists.freeswitch.org
> Sent: Monday, February 02, 2009 9:12 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Freeswitch-users Digest, Vol 32, Issue 17
>
> Send Freeswitch-users mailing list submissions to
>       freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
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> or, via email, send a message with subject or body 'help' to
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> When replying, please edit your Subject line so it is more
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>
>
> Today's Topics:
>
>    1. Re: Call Variable not available when call hangup (shehzad p)
>    2. Re: How do I set my FS internal ip address to a "static"
>       value. (clif at eugeneweb.com)
>    3. Re: Call Variable not available when call hangup
>       (Anthony Minessale)
>    4. Re: How do I set my FS internal ip address to a "static"
>       value. (Brian West)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST)
> From: shehzad p <pmhshz at gmail.com>
> Subject: Re: [Freeswitch-users] Call Variable not available when call
>       hangup
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <21791503.post at talk.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
>
>
> one question is that when javascript is being called from
> dial plan, I get the session object already available, It is
> for A leg of channel, So when javascript is called after
> Bridge how can I get the session object for B leg also?
>
>
> Anthony Minessale-2 wrote:
> >
> > the leg you are running the script on is not hungup, the
> other leg of the
> > call is.
> >
> > If it was hungup you would not be executing the script.
> >
> > Asterisk and the h ext and the whole dead-agi thing are all
> poor design
> > showing it's teeth.
> > We do not support anything like it.
> >
> >
> > You can however try this: (see the link below)
> >
> >
> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
> -giving-me-headaches-p21614840.html
> >
> >
> >
> > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz at gmail.com> wrote:
> >
> >>
> >> Is there any settings that when call hangup control can be
> transferred to
> >> another context and these CDR values can be accessible
> there? (just like
> >> in
> >> Asterisk, h extension)
> >>
> >> shehzad p wrote:
> >> >
> >> > Hi all,
> >> >
> >> > I need to process some CDR variables in Dialplan, like
> call duration,
> >> > Answered time etc.
> >> > but when I place info application after bridge, it is
> not listing them
> >> > properly as below:
> >> > ===========================================
> >> > Caller-Channel-Created-Time: [1233573341672157]
> >> > Caller-Channel-Answered-Time: [1233573342712939]
> >> > Caller-Channel-Hangup-Time: [0]
> >> > ==========================================
> >> > Here Hangup time is 0, So how can I find actual values?
> >> >
> >> > --I know that we can use xml_cdr or cdr_csv, but my
> current need is to
> >> get
> >> > those values from dialplan itself so that can be passed to some
> >> script...
> >> >
> >> >
> >> > thanks,
> >> > msp
> >> >
> >>
> >> --
> >> View this message in context:
> >>
> http://www.nabble.com/Call-Variable-not-available-when-call-ha
> ngup-tp21788550p21789152.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> <MSN%3Aanthony_minessale at hotmail.com>
> >
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
> ny.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> >
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
> f%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/Call-Variable-not-available-when-call-ha
> ngup-tp21788550p21791503.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST)
> From: clif at eugeneweb.com
> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
>       to a    "static" value.
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <Pine.LNX.4.62.0902011323230.2342 at redwall>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> Hi Gang,
>
> I've been struggleing with this also. Actually I can get it
> to bind to my
> address, the problem is it randomly drops my calls. :-(
>
> I have a FS running on a box with a static IP and I can start
> a call between
> two extensions and it will go for hours. Then I add anther
> interface say eth0:0
> with a new static IP and reconfigure my phones and FS to use
> that, and the
> calls drop after about 15-20 mins. Though it's pretty random.
>
> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
> freeswitch-1.0.1.
> Here is my /etc/network/interfaces:
>
> # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8)
>
> # The loopback interface
> auto lo
> iface lo inet loopback
>
> # The first network card - this entry was created during the Debian
> installation
> auto eth0 eth0:0
> iface eth0 inet dhcp
> iface eth0:0 inet static
>       address 192.168.0.249
>       netmask 255.255.255.0
>       gateway 192.168.0.254
>
> The only change I made to the FS config is in Vars.xml. I
> added this line close
> to the top:
>
> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>
> Here is the console log of the call being dropped:
>
> freeswitch at archive> sofia status
> API CALL [sofia(status)] output:
>                       Name       Type
>       Data
> State
> ==============================================================
> ===================================
>                   external    profile
> sip:mod_sofia at 67.171.158.226:5080
> RUNNING (0)
>                   internal    profile
> sip:mod_sofia at 192.168.0.249:5060
> RUNNING (2)
>                        nat    profile
> sip:mod_sofia at 67.171.158.226:5070
> RUNNING (0)
>                    default      alias
>   internal
> ALIASED
>                   outbound      alias
>   external
> ALIASED
>              192.168.0.249      alias
>   internal
> ALIASED
> ==============================================================
> ===================================
> 3 profiles 3 aliases
>
> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
> sofia_glue_restart_all_profiles() Reload XML [Success]
> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler()
> ENUM Reloaded
> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
> sofia_read_frame() Hangup
> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
> ;fs_nat=yes
> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
> switch_ivr_multi_threaded_bridge() Hangup
> sofia/internal/1001 at 192.168.0.249
> [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
> switch_core_session_thread() Session 6
> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud
> p;fs_nat=yes)
> Ended
> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
> switch_core_session_thread() Close Channel
> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
> ;fs_nat=yes
> [CS_HANGUP]
> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
> switch_core_session_thread() Session 5
> (sofia/internal/1001 at 192.168.0.249)
> Ended
> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
> switch_core_session_thread() Close Channel
> sofia/internal/1001 at 192.168.0.249
> [CS_HANGUP]
> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
> sofia_profile_thread_run() waiting for
> worker thread
> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
> sofia_profile_thread_run() waiting for
> worker thread
> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias
> [192.168.0.249] for profile [internal]
> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
> Adding Alias [default]
> for profile [internal]
> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia()
> Started Profile
> internal [sofia_reg_internal]
> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias
> [outbound] for profile [external]
> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
> Started Profile
> external [sofia_reg_external]
> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
> sofia_profile_thread_run() waiting for
> worker thread
> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
> Started Profile nat
> [sofia_reg_nat]
> sofia status
> API CALL [sofia(status)] output:
>                       Name       Type
>       Data
> State
> ==============================================================
> ===================================
>                   external    profile
> sip:mod_sofia at 67.171.158.226:5080
> RUNNING (0)
>                   internal    profile
> sip:mod_sofia at 192.168.0.249:5060
> RUNNING (0)
>                   outbound      alias
>   external
> ALIASED
>              192.168.0.249      alias
>   internal
> ALIASED
>                        nat    profile
> sip:mod_sofia at 67.171.158.226:5070
> RUNNING (0)
>                    default      alias
>   internal
> ALIASED
> ==============================================================
> ===================================
> 3 profiles 3 aliases
>
> There is an older thread that says one should set
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
> but in this (later) thread is says only Jingleling usese that
> variable.
> ie. see:
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
> org/msg00695.html
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
> org/msg07345.html
>
> So what do you think causes this? What is the correct way? ;-)
>
>
>       Thanks,
>       Clif
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 2 Feb 2009 09:41:05 -0600
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] Call Variable not available when call
>       hangup
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>       <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> you can't that's why i said it was a horrible approach.
> That's also why i posted you the instructions on the only
> elegant solution
> to your problem.
>
>
> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p <pmhshz at gmail.com> wrote:
>
> >
> >
> > one question is that when javascript is being called from
> dial plan, I get
> > the session object already available, It is for A leg of channel,
> > So when javascript is called after Bridge how can I get the
> session object
> > for B leg also?
> >
> >
> > Anthony Minessale-2 wrote:
> > >
> > > the leg you are running the script on is not hungup, the
> other leg of the
> > > call is.
> > >
> > > If it was hungup you would not be executing the script.
> > >
> > > Asterisk and the h ext and the whole dead-agi thing are
> all poor design
> > > showing it's teeth.
> > > We do not support anything like it.
> > >
> > >
> > > You can however try this: (see the link below)
> > >
> > >
> >
> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
> -giving-me-headaches-p21614840.html
> > >
> > >
> > >
> > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
> <pmhshz at gmail.com> wrote:
> > >
> > >>
> > >> Is there any settings that when call hangup control can
> be transferred
> > to
> > >> another context and these CDR values can be accessible
> there? (just like
> > >> in
> > >> Asterisk, h extension)
> > >>
> > >> shehzad p wrote:
> > >> >
> > >> > Hi all,
> > >> >
> > >> > I need to process some CDR variables in Dialplan, like
> call duration,
> > >> > Answered time etc.
> > >> > but when I place info application after bridge, it is
> not listing them
> > >> > properly as below:
> > >> > ===========================================
> > >> > Caller-Channel-Created-Time: [1233573341672157]
> > >> > Caller-Channel-Answered-Time: [1233573342712939]
> > >> > Caller-Channel-Hangup-Time: [0]
> > >> > ==========================================
> > >> > Here Hangup time is 0, So how can I find actual values?
> > >> >
> > >> > --I know that we can use xml_cdr or cdr_csv, but my
> current need is to
> > >> get
> > >> > those values from dialplan itself so that can be passed to some
> > >> script...
> > >> >
> > >> >
> > >> > thanks,
> > >> > msp
> > >> >
> > >>
> > >> --
> > >> View this message in context:
> > >>
> >
> http://www.nabble.com/Call-Variable-not-available-when-call-ha
> ngup-tp21788550p21789152.html
> > >> Sent from the Freeswitch-users mailing list archive at
> Nabble.com.
> > >>
> > >>
> > >> _______________________________________________
> > >> Freeswitch-users mailing list
> > >> Freeswitch-users at 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/
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com
> <MSN%3Aanthony_minessale at hotmail.com><
> >
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale@
> hotmail.com>
> > >
> > >
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
> ny.minessale at gmail.com>
> >
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.mines
> sale at gmail.com>
> > >
> > > IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org
> <sip%3A888 at conference.freeswitch.org><
> >
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.fre
> eswitch.org>
> > >
> > > iax:guest at conference.freeswitch.org/888
> > >
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
> f%2B888 at conference.freeswitch.org>
> >
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%
> 253Aconf%252B888 at conference.freeswitch.org>
> > >
> > > pstn:213-799-1400
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> > > http://www.freeswitch.org
> > >
> > >
> >
> > --
> > View this message in context:
> >
> http://www.nabble.com/Call-Variable-not-available-when-call-ha
> ngup-tp21788550p21791503.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
> ny.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
> f%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm
> ents/20090202/2d430e44/attachment-0001.html
>
> ------------------------------
>
> Message: 4
> Date: Mon, 2 Feb 2009 09:41:39 -0600
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
>       to a    "static" value.
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> you need to add this setting to sofia.conf.xml
>
> <param name="auto-restart" value="false"/>
>
>
> You'll also need to edit the sofia profiles and input the
> exact IP you
> wish it to bind to.  The params are sip-ip and rtp-ip.
>
> /b
>
> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote:
>
> > Hi Gang,
> >
> > I've been struggleing with this also. Actually I can get it
> to bind
> > to my
> > address, the problem is it randomly drops my calls. :-(
> >
> > I have a FS running on a box with a static IP and I can
> start a call
> > between
> > two extensions and it will go for hours. Then I add anther
> interface
> > say eth0:0
> > with a new static IP and reconfigure my phones and FS to use that,
> > and the
> > calls drop after about 15-20 mins. Though it's pretty random.
> >
> > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
> > freeswitch-1.0.1.
> > Here is my /etc/network/interfaces:
> >
> > # /etc/network/interfaces -- configuration file for
> ifup(8), ifdown(8)
> >
> > # The loopback interface
> > auto lo
> > iface lo inet loopback
> >
> > # The first network card - this entry was created during the Debian
> > installation
> > auto eth0 eth0:0
> > iface eth0 inet dhcp
> > iface eth0:0 inet static
> >     address 192.168.0.249
> >     netmask 255.255.255.0
> >     gateway 192.168.0.254
> >
> > The only change I made to the FS config is in Vars.xml. I
> added this
> > line close
> > to the top:
> >
> > <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
> >
> > Here is the console log of the call being dropped:
> >
> > freeswitch at archive> sofia status
> > API CALL [sofia(status)] output:
> >                      Name      Type
>       Data
> > State
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> >
> ======================================================================
> >                  external   profile
> sip:mod_sofia at 67.171.158.226:5080
> > RUNNING (0)
> >                  internal   profile
> sip:mod_sofia at 192.168.0.249:5060
> > RUNNING (2)
> >                       nat   profile
> sip:mod_sofia at 67.171.158.226:5070
> > RUNNING (0)
> >                   default     alias
>   internal
> > ALIASED
> >                  outbound     alias
>   external
> > ALIASED
> >             192.168.0.249     alias
>   internal
> > ALIASED
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> >
> ======================================================================
> > 3 profiles 3 aliases
> >
> > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
> > sofia_glue_restart_all_profiles() Reload XML [Success]
> > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM
> > Reloaded
> > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
> sofia_read_frame() Hangup
> > sofia/internal/
> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
> > switch_ivr_multi_threaded_bridge() Hangup
> sofia/internal/1001 at 192.168.0.249
> > [CS_EXECUTE] [NORMAL_CLEARING]
> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
> > switch_core_session_thread() Session 6
> > (sofia/internal/
> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes)
> > Ended
> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
> > switch_core_session_thread() Close Channel
> > sofia/internal/
> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
> > [CS_HANGUP]
> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
> > switch_core_session_thread() Session 5
> (sofia/internal/1001 at 192.168.0.249
> > )
> > Ended
> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
> > switch_core_session_thread() Close Channel
> sofia/internal/1001 at 192.168.0.249
> > [CS_HANGUP]
> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645
> sofia_profile_thread_run()
> > waiting for
> > worker thread
> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645
> sofia_profile_thread_run()
> > waiting for
> > worker thread
> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
> Adding Alias
> > [192.168.0.249] for profile [internal]
> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding
> > Alias [default]
> > for profile [internal]
> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started
> > Profile
> > internal [sofia_reg_internal]
> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia()
> Adding Alias
> > [outbound] for profile [external]
> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
> > Profile
> > external [sofia_reg_external]
> > 2009-02-01 13:23:20 [NOTICE] sofia.c:645
> sofia_profile_thread_run()
> > waiting for
> > worker thread
> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
> > Profile nat
> > [sofia_reg_nat]
> > sofia status
> > API CALL [sofia(status)] output:
> >                      Name      Type
>       Data
> > State
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> >
> ======================================================================
> >                  external   profile
> sip:mod_sofia at 67.171.158.226:5080
> > RUNNING (0)
> >                  internal   profile
> sip:mod_sofia at 192.168.0.249:5060
> > RUNNING (0)
> >                  outbound     alias
>   external
> > ALIASED
> >             192.168.0.249     alias
>   internal
> > ALIASED
> >                       nat   profile
> sip:mod_sofia at 67.171.158.226:5070
> > RUNNING (0)
> >                   default     alias
>   internal
> > ALIASED
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> > =
> >
> ======================================================================
> > 3 profiles 3 aliases
> >
> > There is an older thread that says one should set
> > <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
> > but in this (later) thread is says only Jingleling usese that
> > variable.
> > ie. see:
> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
> org/msg00695.html
> >
> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
> org/msg07345.html
> >
> > So what do you think causes this? What is the correct way? ;-)
> >
> >
> >     Thanks,
> >     Clif
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> > http://www.freeswitch.org
>
>
>
>
> ------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
> itch-users
> http://www.freeswitch.org
>
>
> End of Freeswitch-users Digest, Vol 32, Issue 17
> ************************************************
>


From gmaruzz at celliax.org  Mon Mar  2 04:11:26 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 2 Mar 2009 13:11:26 +0100
Subject: [Freeswitch-users] Running freeswitch on powerpc
In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>
References: <mailman.4507.1233589307.18013.freeswitch-users@lists.freeswitch.org>
	<9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>
Message-ID: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com>

On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
<sridhart at alcatel-lucent.com> wrote:
> I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code

Hi Sridhar,

I don't think someone has tried that. It will probably be you that let
us all know which (if any) changes needs to be done. :-)


Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
<sridhart at alcatel-lucent.com> wrote:
> Hi all,
>
> I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code
>
> Regards
> Sridhar
>
>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
>> Behalf Of freeswitch-users-request at lists.freeswitch.org
>> Sent: Monday, February 02, 2009 9:12 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Freeswitch-users Digest, Vol 32, Issue 17
>>
>> 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: Call Variable not available when call hangup (shehzad p)
>>    2. Re: How do I set my FS internal ip address to a "static"
>>       value. (clif at eugeneweb.com)
>>    3. Re: Call Variable not available when call hangup
>>       (Anthony Minessale)
>>    4. Re: How do I set my FS internal ip address to a "static"
>>       value. (Brian West)
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST)
>> From: shehzad p <pmhshz at gmail.com>
>> Subject: Re: [Freeswitch-users] Call Variable not available when call
>>       hangup
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID: <21791503.post at talk.nabble.com>
>> Content-Type: text/plain; charset=us-ascii
>>
>>
>>
>> one question is that when javascript is being called from
>> dial plan, I get the session object already available, It is
>> for A leg of channel, So when javascript is called after
>> Bridge how can I get the session object for B leg also?
>>
>>
>> Anthony Minessale-2 wrote:
>> >
>> > the leg you are running the script on is not hungup, the
>> other leg of the
>> > call is.
>> >
>> > If it was hungup you would not be executing the script.
>> >
>> > Asterisk and the h ext and the whole dead-agi thing are all
>> poor design
>> > showing it's teeth.
>> > We do not support anything like it.
>> >
>> >
>> > You can however try this: (see the link below)
>> >
>> >
>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>> -giving-me-headaches-p21614840.html
>> >
>> >
>> >
>> > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz at gmail.com> wrote:
>> >
>> >>
>> >> Is there any settings that when call hangup control can be
>> transferred to
>> >> another context and these CDR values can be accessible
>> there? (just like
>> >> in
>> >> Asterisk, h extension)
>> >>
>> >> shehzad p wrote:
>> >> >
>> >> > Hi all,
>> >> >
>> >> > I need to process some CDR variables in Dialplan, like
>> call duration,
>> >> > Answered time etc.
>> >> > but when I place info application after bridge, it is
>> not listing them
>> >> > properly as below:
>> >> > ===========================================
>> >> > Caller-Channel-Created-Time: [1233573341672157]
>> >> > Caller-Channel-Answered-Time: [1233573342712939]
>> >> > Caller-Channel-Hangup-Time: [0]
>> >> > ==========================================
>> >> > Here Hangup time is 0, So how can I find actual values?
>> >> >
>> >> > --I know that we can use xml_cdr or cdr_csv, but my
>> current need is to
>> >> get
>> >> > those values from dialplan itself so that can be passed to some
>> >> script...
>> >> >
>> >> >
>> >> > thanks,
>> >> > msp
>> >> >
>> >>
>> >> --
>> >> View this message in context:
>> >>
>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>> ngup-tp21788550p21789152.html
>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >>
>> >>
>> >> _______________________________________________
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com>
>> >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>> ny.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>> f%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>> ngup-tp21788550p21791503.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>>
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST)
>> From: clif at eugeneweb.com
>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
>>       to a    "static" value.
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID: <Pine.LNX.4.62.0902011323230.2342 at redwall>
>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>>
>> Hi Gang,
>>
>> I've been struggleing with this also. Actually I can get it
>> to bind to my
>> address, the problem is it randomly drops my calls. :-(
>>
>> I have a FS running on a box with a static IP and I can start
>> a call between
>> two extensions and it will go for hours. Then I add anther
>> interface say eth0:0
>> with a new static IP and reconfigure my phones and FS to use
>> that, and the
>> calls drop after about 15-20 mins. Though it's pretty random.
>>
>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>> freeswitch-1.0.1.
>> Here is my /etc/network/interfaces:
>>
>> # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8)
>>
>> # The loopback interface
>> auto lo
>> iface lo inet loopback
>>
>> # The first network card - this entry was created during the Debian
>> installation
>> auto eth0 eth0:0
>> iface eth0 inet dhcp
>> iface eth0:0 inet static
>>       address 192.168.0.249
>>       netmask 255.255.255.0
>>       gateway 192.168.0.254
>>
>> The only change I made to the FS config is in Vars.xml. I
>> added this line close
>> to the top:
>>
>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>>
>> Here is the console log of the call being dropped:
>>
>> freeswitch at archive> sofia status
>> API CALL [sofia(status)] output:
>>                       Name       Type
>>       Data
>> State
>> ==============================================================
>> ===================================
>>                   external    profile
>> sip:mod_sofia at 67.171.158.226:5080
>> RUNNING (0)
>>                   internal    profile
>> sip:mod_sofia at 192.168.0.249:5060
>> RUNNING (2)
>>                        nat    profile
>> sip:mod_sofia at 67.171.158.226:5070
>> RUNNING (0)
>>                    default      alias
>>   internal
>> ALIASED
>>                   outbound      alias
>>   external
>> ALIASED
>>              192.168.0.249      alias
>>   internal
>> ALIASED
>> ==============================================================
>> ===================================
>> 3 profiles 3 aliases
>>
>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>> sofia_glue_restart_all_profiles() Reload XML [Success]
>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler()
>> ENUM Reloaded
>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>> sofia_read_frame() Hangup
>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>> ;fs_nat=yes
>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>> switch_ivr_multi_threaded_bridge() Hangup
>> sofia/internal/1001 at 192.168.0.249
>> [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>> switch_core_session_thread() Session 6
>> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud
>> p;fs_nat=yes)
>> Ended
>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>> switch_core_session_thread() Close Channel
>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>> ;fs_nat=yes
>> [CS_HANGUP]
>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>> switch_core_session_thread() Session 5
>> (sofia/internal/1001 at 192.168.0.249)
>> Ended
>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>> switch_core_session_thread() Close Channel
>> sofia/internal/1001 at 192.168.0.249
>> [CS_HANGUP]
>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>> sofia_profile_thread_run() waiting for
>> worker thread
>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>> sofia_profile_thread_run() waiting for
>> worker thread
>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias
>> [192.168.0.249] for profile [internal]
>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>> Adding Alias [default]
>> for profile [internal]
>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia()
>> Started Profile
>> internal [sofia_reg_internal]
>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias
>> [outbound] for profile [external]
>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>> Started Profile
>> external [sofia_reg_external]
>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>> sofia_profile_thread_run() waiting for
>> worker thread
>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>> Started Profile nat
>> [sofia_reg_nat]
>> sofia status
>> API CALL [sofia(status)] output:
>>                       Name       Type
>>       Data
>> State
>> ==============================================================
>> ===================================
>>                   external    profile
>> sip:mod_sofia at 67.171.158.226:5080
>> RUNNING (0)
>>                   internal    profile
>> sip:mod_sofia at 192.168.0.249:5060
>> RUNNING (0)
>>                   outbound      alias
>>   external
>> ALIASED
>>              192.168.0.249      alias
>>   internal
>> ALIASED
>>                        nat    profile
>> sip:mod_sofia at 67.171.158.226:5070
>> RUNNING (0)
>>                    default      alias
>>   internal
>> ALIASED
>> ==============================================================
>> ===================================
>> 3 profiles 3 aliases
>>
>> There is an older thread that says one should set
>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>> but in this (later) thread is says only Jingleling usese that
>> variable.
>> ie. see:
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>> org/msg00695.html
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>> org/msg07345.html
>>
>> So what do you think causes this? What is the correct way? ;-)
>>
>>
>>       Thanks,
>>       Clif
>>
>>
>>
>>
>> ------------------------------
>>
>> Message: 3
>> Date: Mon, 2 Feb 2009 09:41:05 -0600
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] Call Variable not available when call
>>       hangup
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID:
>>       <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> you can't that's why i said it was a horrible approach.
>> That's also why i posted you the instructions on the only
>> elegant solution
>> to your problem.
>>
>>
>> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p <pmhshz at gmail.com> wrote:
>>
>> >
>> >
>> > one question is that when javascript is being called from
>> dial plan, I get
>> > the session object already available, It is for A leg of channel,
>> > So when javascript is called after Bridge how can I get the
>> session object
>> > for B leg also?
>> >
>> >
>> > Anthony Minessale-2 wrote:
>> > >
>> > > the leg you are running the script on is not hungup, the
>> other leg of the
>> > > call is.
>> > >
>> > > If it was hungup you would not be executing the script.
>> > >
>> > > Asterisk and the h ext and the whole dead-agi thing are
>> all poor design
>> > > showing it's teeth.
>> > > We do not support anything like it.
>> > >
>> > >
>> > > You can however try this: (see the link below)
>> > >
>> > >
>> >
>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>> -giving-me-headaches-p21614840.html
>> > >
>> > >
>> > >
>> > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
>> <pmhshz at gmail.com> wrote:
>> > >
>> > >>
>> > >> Is there any settings that when call hangup control can
>> be transferred
>> > to
>> > >> another context and these CDR values can be accessible
>> there? (just like
>> > >> in
>> > >> Asterisk, h extension)
>> > >>
>> > >> shehzad p wrote:
>> > >> >
>> > >> > Hi all,
>> > >> >
>> > >> > I need to process some CDR variables in Dialplan, like
>> call duration,
>> > >> > Answered time etc.
>> > >> > but when I place info application after bridge, it is
>> not listing them
>> > >> > properly as below:
>> > >> > ===========================================
>> > >> > Caller-Channel-Created-Time: [1233573341672157]
>> > >> > Caller-Channel-Answered-Time: [1233573342712939]
>> > >> > Caller-Channel-Hangup-Time: [0]
>> > >> > ==========================================
>> > >> > Here Hangup time is 0, So how can I find actual values?
>> > >> >
>> > >> > --I know that we can use xml_cdr or cdr_csv, but my
>> current need is to
>> > >> get
>> > >> > those values from dialplan itself so that can be passed to some
>> > >> script...
>> > >> >
>> > >> >
>> > >> > thanks,
>> > >> > msp
>> > >> >
>> > >>
>> > >> --
>> > >> View this message in context:
>> > >>
>> >
>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>> ngup-tp21788550p21789152.html
>> > >> Sent from the Freeswitch-users mailing list archive at
>> Nabble.com.
>> > >>
>> > >>
>> > >> _______________________________________________
>> > >> Freeswitch-users mailing list
>> > >> Freeswitch-users at 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/
>> > >
>> > > AIM: anthm
>> > > MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com><
>> >
>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale@
>> hotmail.com>
>> > >
>> > >
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>> ny.minessale at gmail.com>
>> >
>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.mines
>> sale at gmail.com>
>> > >
>> > > IRC: irc.freenode.net #freeswitch
>> > >
>> > > FreeSWITCH Developer Conference
>> > > sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org><
>> >
>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.fre
>> eswitch.org>
>> > >
>> > > iax:guest at conference.freeswitch.org/888
>> > >
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>> f%2B888 at conference.freeswitch.org>
>> >
>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%
>> 253Aconf%252B888 at conference.freeswitch.org>
>> > >
>> > > pstn:213-799-1400
>> > >
>> > > _______________________________________________
>> > > Freeswitch-users mailing list
>> > > Freeswitch-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> > > http://www.freeswitch.org
>> > >
>> > >
>> >
>> > --
>> > View this message in context:
>> >
>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>> ngup-tp21788550p21791503.html
>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>> ny.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>> f%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL:
>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm
>> ents/20090202/2d430e44/attachment-0001.html
>>
>> ------------------------------
>>
>> Message: 4
>> Date: Mon, 2 Feb 2009 09:41:39 -0600
>> From: Brian West <brian at freeswitch.org>
>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip address
>>       to a    "static" value.
>> To: freeswitch-users at lists.freeswitch.org
>> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org>
>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>>
>> you need to add this setting to sofia.conf.xml
>>
>> <param name="auto-restart" value="false"/>
>>
>>
>> You'll also need to edit the sofia profiles and input the
>> exact IP you
>> wish it to bind to.  The params are sip-ip and rtp-ip.
>>
>> /b
>>
>> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote:
>>
>> > Hi Gang,
>> >
>> > I've been struggleing with this also. Actually I can get it
>> to bind
>> > to my
>> > address, the problem is it randomly drops my calls. :-(
>> >
>> > I have a FS running on a box with a static IP and I can
>> start a call
>> > between
>> > two extensions and it will go for hours. Then I add anther
>> interface
>> > say eth0:0
>> > with a new static IP and reconfigure my phones and FS to use that,
>> > and the
>> > calls drop after about 15-20 mins. Though it's pretty random.
>> >
>> > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>> > freeswitch-1.0.1.
>> > Here is my /etc/network/interfaces:
>> >
>> > # /etc/network/interfaces -- configuration file for
>> ifup(8), ifdown(8)
>> >
>> > # The loopback interface
>> > auto lo
>> > iface lo inet loopback
>> >
>> > # The first network card - this entry was created during the Debian
>> > installation
>> > auto eth0 eth0:0
>> > iface eth0 inet dhcp
>> > iface eth0:0 inet static
>> >     address 192.168.0.249
>> >     netmask 255.255.255.0
>> >     gateway 192.168.0.254
>> >
>> > The only change I made to the FS config is in Vars.xml. I
>> added this
>> > line close
>> > to the top:
>> >
>> > <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>> >
>> > Here is the console log of the call being dropped:
>> >
>> > freeswitch at archive> sofia status
>> > API CALL [sofia(status)] output:
>> >                      Name      Type
>>       Data
>> > State
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> >
>> ======================================================================
>> >                  external   profile
>> sip:mod_sofia at 67.171.158.226:5080
>> > RUNNING (0)
>> >                  internal   profile
>> sip:mod_sofia at 192.168.0.249:5060
>> > RUNNING (2)
>> >                       nat   profile
>> sip:mod_sofia at 67.171.158.226:5070
>> > RUNNING (0)
>> >                   default     alias
>>   internal
>> > ALIASED
>> >                  outbound     alias
>>   external
>> > ALIASED
>> >             192.168.0.249     alias
>>   internal
>> > ALIASED
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> >
>> ======================================================================
>> > 3 profiles 3 aliases
>> >
>> > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>> > sofia_glue_restart_all_profiles() Reload XML [Success]
>> > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM
>> > Reloaded
>> > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>> sofia_read_frame() Hangup
>> > sofia/internal/
>> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>> > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>> > switch_ivr_multi_threaded_bridge() Hangup
>> sofia/internal/1001 at 192.168.0.249
>> > [CS_EXECUTE] [NORMAL_CLEARING]
>> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>> > switch_core_session_thread() Session 6
>> > (sofia/internal/
>> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes)
>> > Ended
>> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>> > switch_core_session_thread() Close Channel
>> > sofia/internal/
>> > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>> > [CS_HANGUP]
>> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>> > switch_core_session_thread() Session 5
>> (sofia/internal/1001 at 192.168.0.249
>> > )
>> > Ended
>> > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>> > switch_core_session_thread() Close Channel
>> sofia/internal/1001 at 192.168.0.249
>> > [CS_HANGUP]
>> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>> sofia_profile_thread_run()
>> > waiting for
>> > worker thread
>> > 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>> sofia_profile_thread_run()
>> > waiting for
>> > worker thread
>> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>> Adding Alias
>> > [192.168.0.249] for profile [internal]
>> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding
>> > Alias [default]
>> > for profile [internal]
>> > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started
>> > Profile
>> > internal [sofia_reg_internal]
>> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia()
>> Adding Alias
>> > [outbound] for profile [external]
>> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>> > Profile
>> > external [sofia_reg_external]
>> > 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>> sofia_profile_thread_run()
>> > waiting for
>> > worker thread
>> > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>> > Profile nat
>> > [sofia_reg_nat]
>> > sofia status
>> > API CALL [sofia(status)] output:
>> >                      Name      Type
>>       Data
>> > State
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> >
>> ======================================================================
>> >                  external   profile
>> sip:mod_sofia at 67.171.158.226:5080
>> > RUNNING (0)
>> >                  internal   profile
>> sip:mod_sofia at 192.168.0.249:5060
>> > RUNNING (0)
>> >                  outbound     alias
>>   external
>> > ALIASED
>> >             192.168.0.249     alias
>>   internal
>> > ALIASED
>> >                       nat   profile
>> sip:mod_sofia at 67.171.158.226:5070
>> > RUNNING (0)
>> >                   default     alias
>>   internal
>> > ALIASED
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> > =
>> >
>> ======================================================================
>> > 3 profiles 3 aliases
>> >
>> > There is an older thread that says one should set
>> > <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>> > but in this (later) thread is says only Jingleling usese that
>> > variable.
>> > ie. see:
>> >
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>> org/msg00695.html
>> >
>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>> org/msg07345.html
>> >
>> > So what do you think causes this? What is the correct way? ;-)
>> >
>> >
>> >     Thanks,
>> >     Clif
>> >
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> > http://www.freeswitch.org
>>
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>> itch-users
>> http://www.freeswitch.org
>>
>>
>> End of Freeswitch-users Digest, Vol 32, Issue 17
>> ************************************************
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 wojtek at VoIPMan.ORG  Mon Mar  2 04:32:31 2009
From: wojtek at VoIPMan.ORG (Wojciech Tryc)
Date: Mon, 2 Mar 2009 07:32:31 -0500
Subject: [Freeswitch-users] Running freeswitch on powerpc
In-Reply-To: <7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com>
References: <mailman.4507.1233589307.18013.freeswitch-users@lists.freeswitch.org>
	<9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>
	<7b197bef0903020411q72be83efxace263ab3401c001@mail.gmail.com>
Message-ID: <60097A1C-2820-4397-BBEE-141FC7FEE3AE@VoIPMan.ORG>

Sridhar,
PIKA's WARP is PowerPC based...AMCC but still Big Endian and PowerPC.
 From what I remember the endianness definition was broken in one or  
two places, but other than that it was effortless (native compilation).

Thanks,
Wojtek,

On Mar 2, 2009, at 7:11 AM, Giovanni Maruzzelli wrote:

> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
> <sridhart at alcatel-lucent.com> wrote:
>> I am planning to run freeswitch on powerpc MPC8358. Please let me  
>> know if any changes needs to be done in the code
>
> Hi Sridhar,
>
> I don't think someone has tried that. It will probably be you that let
> us all know which (if any) changes needs to be done. :-)
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Mon, Mar 2, 2009 at 12:52 PM, Rajagopal, Sridhar (Sridhar)
> <sridhart at alcatel-lucent.com> wrote:
>> Hi all,
>>
>> I am planning to run freeswitch on powerpc MPC8358. Please let me  
>> know if any changes needs to be done in the code
>>
>> Regards
>> Sridhar
>>
>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On
>>> Behalf Of freeswitch-users-request at lists.freeswitch.org
>>> Sent: Monday, February 02, 2009 9:12 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Freeswitch-users Digest, Vol 32, Issue 17
>>>
>>> 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: Call Variable not available when call hangup (shehzad p)
>>>   2. Re: How do I set my FS internal ip address to a "static"
>>>      value. (clif at eugeneweb.com)
>>>   3. Re: Call Variable not available when call hangup
>>>      (Anthony Minessale)
>>>   4. Re: How do I set my FS internal ip address to a "static"
>>>      value. (Brian West)
>>>
>>>
>>> ----------------------------------------------------------------------
>>>
>>> Message: 1
>>> Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST)
>>> From: shehzad p <pmhshz at gmail.com>
>>> Subject: Re: [Freeswitch-users] Call Variable not available when  
>>> call
>>>      hangup
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <21791503.post at talk.nabble.com>
>>> Content-Type: text/plain; charset=us-ascii
>>>
>>>
>>>
>>> one question is that when javascript is being called from
>>> dial plan, I get the session object already available, It is
>>> for A leg of channel, So when javascript is called after
>>> Bridge how can I get the session object for B leg also?
>>>
>>>
>>> Anthony Minessale-2 wrote:
>>>>
>>>> the leg you are running the script on is not hungup, the
>>> other leg of the
>>>> call is.
>>>>
>>>> If it was hungup you would not be executing the script.
>>>>
>>>> Asterisk and the h ext and the whole dead-agi thing are all
>>> poor design
>>>> showing it's teeth.
>>>> We do not support anything like it.
>>>>
>>>>
>>>> You can however try this: (see the link below)
>>>>
>>>>
>>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>>> -giving-me-headaches-p21614840.html
>>>>
>>>>
>>>>
>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p <pmhshz at gmail.com> wrote:
>>>>
>>>>>
>>>>> Is there any settings that when call hangup control can be
>>> transferred to
>>>>> another context and these CDR values can be accessible
>>> there? (just like
>>>>> in
>>>>> Asterisk, h extension)
>>>>>
>>>>> shehzad p wrote:
>>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> I need to process some CDR variables in Dialplan, like
>>> call duration,
>>>>>> Answered time etc.
>>>>>> but when I place info application after bridge, it is
>>> not listing them
>>>>>> properly as below:
>>>>>> ===========================================
>>>>>> Caller-Channel-Created-Time: [1233573341672157]
>>>>>> Caller-Channel-Answered-Time: [1233573342712939]
>>>>>> Caller-Channel-Hangup-Time: [0]
>>>>>> ==========================================
>>>>>> Here Hangup time is 0, So how can I find actual values?
>>>>>>
>>>>>> --I know that we can use xml_cdr or cdr_csv, but my
>>> current need is to
>>>>> get
>>>>>> those values from dialplan itself so that can be passed to some
>>>>> script...
>>>>>>
>>>>>>
>>>>>> thanks,
>>>>>> msp
>>>>>>
>>>>>
>>>>> --
>>>>> View this message in context:
>>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21789152.html
>>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com>
>>>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21791503.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 2
>>> Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST)
>>> From: clif at eugeneweb.com
>>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip  
>>> address
>>>      to a    "static" value.
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <Pine.LNX.4.62.0902011323230.2342 at redwall>
>>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>>>
>>> Hi Gang,
>>>
>>> I've been struggleing with this also. Actually I can get it
>>> to bind to my
>>> address, the problem is it randomly drops my calls. :-(
>>>
>>> I have a FS running on a box with a static IP and I can start
>>> a call between
>>> two extensions and it will go for hours. Then I add anther
>>> interface say eth0:0
>>> with a new static IP and reconfigure my phones and FS to use
>>> that, and the
>>> calls drop after about 15-20 mins. Though it's pretty random.
>>>
>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>>> freeswitch-1.0.1.
>>> Here is my /etc/network/interfaces:
>>>
>>> # /etc/network/interfaces -- configuration file for ifup(8),  
>>> ifdown(8)
>>>
>>> # The loopback interface
>>> auto lo
>>> iface lo inet loopback
>>>
>>> # The first network card - this entry was created during the Debian
>>> installation
>>> auto eth0 eth0:0
>>> iface eth0 inet dhcp
>>> iface eth0:0 inet static
>>>      address 192.168.0.249
>>>      netmask 255.255.255.0
>>>      gateway 192.168.0.254
>>>
>>> The only change I made to the FS config is in Vars.xml. I
>>> added this line close
>>> to the top:
>>>
>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>>>
>>> Here is the console log of the call being dropped:
>>>
>>> freeswitch at archive> sofia status
>>> API CALL [sofia(status)] output:
>>>                      Name       Type
>>>      Data
>>> State
>>> ==============================================================
>>> ===================================
>>>                  external    profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>> RUNNING (0)
>>>                  internal    profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>> RUNNING (2)
>>>                       nat    profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>> RUNNING (0)
>>>                   default      alias
>>>  internal
>>> ALIASED
>>>                  outbound      alias
>>>  external
>>> ALIASED
>>>             192.168.0.249      alias
>>>  internal
>>> ALIASED
>>> ==============================================================
>>> ===================================
>>> 3 profiles 3 aliases
>>>
>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>>> sofia_glue_restart_all_profiles() Reload XML [Success]
>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler()
>>> ENUM Reloaded
>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>>> sofia_read_frame() Hangup
>>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>>> ;fs_nat=yes
>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>>> switch_ivr_multi_threaded_bridge() Hangup
>>> sofia/internal/1001 at 192.168.0.249
>>> [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>> switch_core_session_thread() Session 6
>>> (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=ud
>>> p;fs_nat=yes)
>>> Ended
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp
>>> ;fs_nat=yes
>>> [CS_HANGUP]
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>> switch_core_session_thread() Session 5
>>> (sofia/internal/1001 at 192.168.0.249)
>>> Ended
>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1001 at 192.168.0.249
>>> [CS_HANGUP]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding  
>>> Alias
>>> [192.168.0.249] for profile [internal]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias [default]
>>> for profile [internal]
>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile
>>> internal [sofia_reg_internal]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding  
>>> Alias
>>> [outbound] for profile [external]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile
>>> external [sofia_reg_external]
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run() waiting for
>>> worker thread
>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia()
>>> Started Profile nat
>>> [sofia_reg_nat]
>>> sofia status
>>> API CALL [sofia(status)] output:
>>>                      Name       Type
>>>      Data
>>> State
>>> ==============================================================
>>> ===================================
>>>                  external    profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>> RUNNING (0)
>>>                  internal    profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>> RUNNING (0)
>>>                  outbound      alias
>>>  external
>>> ALIASED
>>>             192.168.0.249      alias
>>>  internal
>>> ALIASED
>>>                       nat    profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>> RUNNING (0)
>>>                   default      alias
>>>  internal
>>> ALIASED
>>> ==============================================================
>>> ===================================
>>> 3 profiles 3 aliases
>>>
>>> There is an older thread that says one should set
>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>>> but in this (later) thread is says only Jingleling usese that
>>> variable.
>>> ie. see:
>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>>> org/msg00695.html
>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>>> org/msg07345.html
>>>
>>> So what do you think causes this? What is the correct way? ;-)
>>>
>>>
>>>      Thanks,
>>>      Clif
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> Message: 3
>>> Date: Mon, 2 Feb 2009 09:41:05 -0600
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> Subject: Re: [Freeswitch-users] Call Variable not available when  
>>> call
>>>      hangup
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID:
>>>      <191c3a030902020741k779e2488o38ca578a3b40e9ad at mail.gmail.com>
>>> Content-Type: text/plain; charset="iso-8859-1"
>>>
>>> you can't that's why i said it was a horrible approach.
>>> That's also why i posted you the instructions on the only
>>> elegant solution
>>> to your problem.
>>>
>>>
>>> On Mon, Feb 2, 2009 at 9:21 AM, shehzad p <pmhshz at gmail.com> wrote:
>>>
>>>>
>>>>
>>>> one question is that when javascript is being called from
>>> dial plan, I get
>>>> the session object already available, It is for A leg of channel,
>>>> So when javascript is called after Bridge how can I get the
>>> session object
>>>> for B leg also?
>>>>
>>>>
>>>> Anthony Minessale-2 wrote:
>>>>>
>>>>> the leg you are running the script on is not hungup, the
>>> other leg of the
>>>>> call is.
>>>>>
>>>>> If it was hungup you would not be executing the script.
>>>>>
>>>>> Asterisk and the h ext and the whole dead-agi thing are
>>> all poor design
>>>>> showing it's teeth.
>>>>> We do not support anything like it.
>>>>>
>>>>>
>>>>> You can however try this: (see the link below)
>>>>>
>>>>>
>>>>
>>> http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks
>>> -giving-me-headaches-p21614840.html
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Feb 2, 2009 at 6:53 AM, shehzad p
>>> <pmhshz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>> Is there any settings that when call hangup control can
>>> be transferred
>>>> to
>>>>>> another context and these CDR values can be accessible
>>> there? (just like
>>>>>> in
>>>>>> Asterisk, h extension)
>>>>>>
>>>>>> shehzad p wrote:
>>>>>>>
>>>>>>> Hi all,
>>>>>>>
>>>>>>> I need to process some CDR variables in Dialplan, like
>>> call duration,
>>>>>>> Answered time etc.
>>>>>>> but when I place info application after bridge, it is
>>> not listing them
>>>>>>> properly as below:
>>>>>>> ===========================================
>>>>>>> Caller-Channel-Created-Time: [1233573341672157]
>>>>>>> Caller-Channel-Answered-Time: [1233573342712939]
>>>>>>> Caller-Channel-Hangup-Time: [0]
>>>>>>> ==========================================
>>>>>>> Here Hangup time is 0, So how can I find actual values?
>>>>>>>
>>>>>>> --I know that we can use xml_cdr or cdr_csv, but my
>>> current need is to
>>>>>> get
>>>>>>> those values from dialplan itself so that can be passed to some
>>>>>> script...
>>>>>>>
>>>>>>>
>>>>>>> thanks,
>>>>>>> msp
>>>>>>>
>>>>>>
>>>>>> --
>>>>>> View this message in context:
>>>>>>
>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21789152.html
>>>>>> Sent from the Freeswitch-users mailing list archive at
>>> Nabble.com.
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Freeswitch-users mailing list
>>>>>> Freeswitch-users at 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/
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com><
>>>>
>>> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale@
>>> hotmail.com>
>>>>>
>>>>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>>>
>>> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.mines
>>> sale at gmail.com>
>>>>>
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org><
>>>>
>>> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.fre
>>> eswitch.org>
>>>>>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>>
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>>>
>>> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%
>>> 253Aconf%252B888 at conference.freeswitch.org>
>>>>>
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> --
>>>> View this message in context:
>>>>
>>> http://www.nabble.com/Call-Variable-not-available-when-call-ha
>>> ngup-tp21788550p21791503.html
>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aantho
>>> ny.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Acon
>>> f%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>> -------------- next part --------------
>>> An HTML attachment was scrubbed...
>>> URL:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachm
>>> ents/20090202/2d430e44/attachment-0001.html
>>>
>>> ------------------------------
>>>
>>> Message: 4
>>> Date: Mon, 2 Feb 2009 09:41:39 -0600
>>> From: Brian West <brian at freeswitch.org>
>>> Subject: Re: [Freeswitch-users] How do I set my FS internal ip  
>>> address
>>>      to a    "static" value.
>>> To: freeswitch-users at lists.freeswitch.org
>>> Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C at freeswitch.org>
>>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>>>
>>> you need to add this setting to sofia.conf.xml
>>>
>>> <param name="auto-restart" value="false"/>
>>>
>>>
>>> You'll also need to edit the sofia profiles and input the
>>> exact IP you
>>> wish it to bind to.  The params are sip-ip and rtp-ip.
>>>
>>> /b
>>>
>>> On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote:
>>>
>>>> Hi Gang,
>>>>
>>>> I've been struggleing with this also. Actually I can get it
>>> to bind
>>>> to my
>>>> address, the problem is it randomly drops my calls. :-(
>>>>
>>>> I have a FS running on a box with a static IP and I can
>>> start a call
>>>> between
>>>> two extensions and it will go for hours. Then I add anther
>>> interface
>>>> say eth0:0
>>>> with a new static IP and reconfigure my phones and FS to use that,
>>>> and the
>>>> calls drop after about 15-20 mins. Though it's pretty random.
>>>>
>>>> Here is my setup. I have Debian Linux 2.6.23.1 kernel, and
>>>> freeswitch-1.0.1.
>>>> Here is my /etc/network/interfaces:
>>>>
>>>> # /etc/network/interfaces -- configuration file for
>>> ifup(8), ifdown(8)
>>>>
>>>> # The loopback interface
>>>> auto lo
>>>> iface lo inet loopback
>>>>
>>>> # The first network card - this entry was created during the Debian
>>>> installation
>>>> auto eth0 eth0:0
>>>> iface eth0 inet dhcp
>>>> iface eth0:0 inet static
>>>>    address 192.168.0.249
>>>>    netmask 255.255.255.0
>>>>    gateway 192.168.0.254
>>>>
>>>> The only change I made to the FS config is in Vars.xml. I
>>> added this
>>>> line close
>>>> to the top:
>>>>
>>>> <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.0.249"/>
>>>>
>>>> Here is the console log of the call being dropped:
>>>>
>>>> freeswitch at archive> sofia status
>>>> API CALL [sofia(status)] output:
>>>>                     Name      Type
>>>      Data
>>>> State
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>>                 external   profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>>> RUNNING (0)
>>>>                 internal   profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>>> RUNNING (2)
>>>>                      nat   profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>>> RUNNING (0)
>>>>                  default     alias
>>>  internal
>>>> ALIASED
>>>>                 outbound     alias
>>>  external
>>>> ALIASED
>>>>            192.168.0.249     alias
>>>  internal
>>>> ALIASED
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>> 3 profiles 3 aliases
>>>>
>>>> freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634
>>>> sofia_glue_restart_all_profiles() Reload XML [Success]
>>>> 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM
>>>> Reloaded
>>>> 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568
>>> sofia_read_frame() Hangup
>>>> sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820
>>>> switch_ivr_multi_threaded_bridge() Hangup
>>> sofia/internal/1001 at 192.168.0.249
>>>> [CS_EXECUTE] [NORMAL_CLEARING]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>>> switch_core_session_thread() Session 6
>>>> (sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes)
>>>> Ended
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>>> switch_core_session_thread() Close Channel
>>>> sofia/internal/
>>>> 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes
>>>> [CS_HANGUP]
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807
>>>> switch_core_session_thread() Session 5
>>> (sofia/internal/1001 at 192.168.0.249
>>>> )
>>>> Ended
>>>> 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809
>>>> switch_core_session_thread() Close Channel
>>> sofia/internal/1001 at 192.168.0.249
>>>> [CS_HANGUP]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias
>>>> [192.168.0.249] for profile [internal]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding
>>>> Alias [default]
>>>> for profile [internal]
>>>> 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile
>>>> internal [sofia_reg_internal]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia()
>>> Adding Alias
>>>> [outbound] for profile [external]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile
>>>> external [sofia_reg_external]
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:645
>>> sofia_profile_thread_run()
>>>> waiting for
>>>> worker thread
>>>> 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started
>>>> Profile nat
>>>> [sofia_reg_nat]
>>>> sofia status
>>>> API CALL [sofia(status)] output:
>>>>                     Name      Type
>>>      Data
>>>> State
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>>                 external   profile
>>> sip:mod_sofia at 67.171.158.226:5080
>>>> RUNNING (0)
>>>>                 internal   profile
>>> sip:mod_sofia at 192.168.0.249:5060
>>>> RUNNING (0)
>>>>                 outbound     alias
>>>  external
>>>> ALIASED
>>>>            192.168.0.249     alias
>>>  internal
>>>> ALIASED
>>>>                      nat   profile
>>> sip:mod_sofia at 67.171.158.226:5070
>>>> RUNNING (0)
>>>>                  default     alias
>>>  internal
>>>> ALIASED
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>> =
>>>>
>>> = 
>>> = 
>>> ====================================================================
>>>> 3 profiles 3 aliases
>>>>
>>>> There is an older thread that says one should set
>>>> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.0.249"/>
>>>> but in this (later) thread is says only Jingleling usese that
>>>> variable.
>>>> ie. see:
>>>>
>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>>> org/msg00695.html
>>>>
>>> http://www.mail-archive.com/freeswitch-users at lists.freeswitch.
>>> org/msg07345.html
>>>>
>>>> So what do you think causes this? What is the correct way? ;-)
>>>>
>>>>
>>>>    Thanks,
>>>>    Clif
>>>>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freesw
>>> itch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> End of Freeswitch-users Digest, Vol 32, Issue 17
>>> ************************************************
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Mon Mar  2 04:50:29 2009
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 02 Mar 2009 20:50:29 +0800
Subject: [Freeswitch-users] Running freeswitch on powerpc
In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>
References: <mailman.4507.1233589307.18013.freeswitch-users@lists.freeswitch.org>
	<9389DD3DDD6B9144B147CE564C6599B902D176FA93@INBANSXCHMBSA3.in.alcatel-lucent.com>
Message-ID: <49ABD615.9050906@coppice.org>

Rajagopal, Sridhar (Sridhar) wrote:
> Hi all,
>
> I am planning to run freeswitch on powerpc MPC8358. Please let me know if any changes needs to be done in the code
>
> Regards
> Sridhar
>   
It may be easier to say what will currently stop Freeswitch working.

The lack of an MMU is a problem right now, so Blackfins are out, which 
is sad. Cores without hardware floating point may not perform all that 
well, but should work. Endianness should not be a problem. I think 
machines which choke on misaligned access are probably OK, too.

Checking that list, you should be OK on a PPC.

Steve



From anthony.minessale at gmail.com  Mon Mar  2 05:50:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 07:50:32 -0600
Subject: [Freeswitch-users] ESL Wrapper
In-Reply-To: <2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<49A6E1DB.3070806@freeswitch.org>
	<1235740392995-2395557.post@n2.nabble.com>
	<2ea4d47e0903020129y676bbc35n8727f8906237edce@mail.gmail.com>
Message-ID: <191c3a030903020550x44ee80e3tcf33b805c7c30d5e@mail.gmail.com>

pardon?
ESL is just a client library for event socket to make it easier to make
event socket apps.
ESL == Event Socket Library


On Mon, Mar 2, 2009 at 3:29 AM, Gopal krishnan <gopal2krishnan at gmail.com>wrote:

> Hi,
>   Actually what is the difference between ESL in FS 1.0.3 and event socket
> in FS 1.0.2. Is the FS 1.0.3 ESL superior?
>
> On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex <rex.alex345 at yahoo.com> wrote:
>
>> Hi All, I did what you have all suggested. Now its working perfectly.
>> Thanks a lot for all your assistance. Rex.
>>
>> Raymond Chandler wrote:
>>  and it will probably be a good idea to do make phpmod-install so that the
>> .so and .php files gets into the correct place to be included -Ray Mathieu
>> Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at
>> 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh &&
>> ./configure && make >> install and did Mathieu's suggestion but getting
>> error as below, >> [root at server esl]# make phpmod make
>> MYLIB="../libesl.a" >> SOLINK="-shared -Xlinker -x" >>
>> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g
>> -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror
>> -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>
>> -Wmissing-prototypes" >>
>> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE
>> -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]:
>> php-config: Command not found make[1]: >> Entering directory
>> `/root/freeswitch-1.0.3/libs/esl/php' g++ >>
>> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb
>> -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >>
>> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >>
>> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >>
>> esl_wrap.cpp:719:17: error: php.h: No such file or directory >>
>> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >>
>> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >>
>> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >>
>> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >>
>> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >>
>> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error:
>> expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or
>> field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error:
>> ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was
>> not declared in this scope >> esl_wrap.cpp:793: error: expected
>> primary-expression before ?void? >> esl_wrap.cpp:793: error: expected
>> primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was
>> not declared in this scope >> esl_wrap.cpp:793: error: expected
>> primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer
>> expression list treated as >> compound expression esl_wrap.cpp:793: error:
>> expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1
>> make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make:
>> *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i
>> wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34
>> AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it..
>> Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh &&
>> ./configure && >> make install And then do Mathieu's suggestion? -MC >>
>> _______________________________________________ Freeswitch-users >> mailing
>> list Freeswitch-users at lists.freeswitch.org >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>>
>> http://www.freeswitch.org >> >> >>
>> ------------------------------------------------------------------------ >>
>> View this message in context: Re: ESL Wrapper >> >> Sent from the
>> freeswitch-users mailing list archive >> at Nabble.com. >>
>> _______________________________________________ >> Freeswitch-users mailing
>> list >> Freeswitch-users at lists.freeswitch.org >> >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>>
>> http://www.freeswitch.org > >
>> ------------------------------------------------------------------------ > >
>> _______________________________________________ > Freeswitch-users mailing
>> list > Freeswitch-users at lists.freeswitch.org >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org >
>> _______________________________________________ Freeswitch-users mailing
>> list Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> View this message in context: Re: ESL Wrapper<http://n2.nabble.com/ESL-Wrapper-tp2385651p2395557.html>
>> Sent from the freeswitch-users mailing list archive<http://n2.nabble.com/freeswitch-users-f2379917.html>at Nabble.com.
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thank you  with regards,
> Gopal,
> PeopleTech Systems Private Limited
> www.peopletech.co.in
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
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From anthony.minessale at gmail.com  Mon Mar  2 05:56:50 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 07:56:50 -0600
Subject: [Freeswitch-users] hide caller_id_number,
	transfer and then 	change caller_id_number
In-Reply-To: <49AB004C.6090604@gmx.net>
References: <49AB004C.6090604@gmx.net>
Message-ID: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>

put origination_caller_id_number in the dial string of any call and you can
set the caller id individually for that leg

{origination_caller_id_number=1234}<any normal dial string>


On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> I have the following problem while providing callback (mod_eventsocket
> is used):
> 1) I want to call a certain destination number A with a suppressed
> caller_id_number (this works fine with some vars in the origination string)
> 2) The destination number A picks up the phone and enters a target
> number B by DTMF
> 3) freeswitch then forwards the call to target number B by DTMF and I
> want to show the number A. I do this with uuid_setvar. The problem is
> that it still shows unknown.
> This is all with SIP.
>
> uuid_setvar however worked if I did not set the caller_id_number to
> unknown. Per default this is then "00000000000" and can then be changed
> with uuid_setvar to the number of A.
> But if I set caller_id_number to unknown I can no longer change it to A.
>
> Any hint?
>
> Best regards
> Peter
>
>
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Mon Mar  2 05:59:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 07:59:34 -0600
Subject: [Freeswitch-users] Orginate: getting status of call fail
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC04@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC04@sun.Noble.local>
Message-ID: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com>

The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the
custom variables to identify what you want to do with the failed calls.



On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?ve been running a test script written in lua which now works very well
> thanks to Anthony?s fix to stream file.
>
>
>
> Right now I?m using an event socket to initiate the call and passing the
> name of the script along with originate thus:
>
>
>
> $dialstring = "originate
> {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum
> '&lua(helloworld.lua )'";
>
> $result = $obj ->bgapi_command($dialstring);
>
>
>
> The script gets fired (it would appear) on answer.  However, if the number
> is invalid , timed out or was busy, I?m not sure the script gets executed or
> am I wrong?
>
>
>
> I want to be able to fire an event back on what happed to the call in the
> event that it failed for whatever reason.
>
>
>
> I know I can simply call the originate and pass the number as an argument
> and execute the dial within the script but I?m led to believe that?s not
> very efficient, or am I completely wrong?
>
>
>
> Looking for the most FS friendly way here
>
>
>
> Regards,
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From nik.middleton at noblesolutions.co.uk  Mon Mar  2 06:49:03 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Mon, 2 Mar 2009 14:49:03 -0000
Subject: [Freeswitch-users] Orginate: getting status of call fail
In-Reply-To: <191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC04@sun.Noble.local>
	<191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BC17@sun.Noble.local>

That's what I was wondering, however, won't the response to the bagi
(not the initial) give me the info on the call result?

 

Regards

 

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 02 March 2009 14:00
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orginate: getting status of call fail

 

The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the 
custom variables to identify what you want to do with the failed calls.




On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton
<nik.middleton at noblesolutions.co.uk> wrote:

Hi Guys,

 

I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.

 

Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:

 

$dialstring = "originate
{ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/
Mygw/phonenum '&lua(helloworld.lua )'";

$result = $obj ->bgapi_command($dialstring);

 

The script gets fired (it would appear) on answer.  However, if the
number is invalid , timed out or was busy, I'm not sure the script gets
executed or am I wrong?

 

I want to be able to fire an event back on what happed to the call in
the event that it failed for whatever reason.

 

I know I can simply call the originate and pass the number as an
argument and execute the dial within the script but I'm led to believe
that's not very efficient, or am I completely wrong?

 

Looking for the most FS friendly way here

 

Regards,


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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/

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> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
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pstn:213-799-1400

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From anthony.minessale at gmail.com  Mon Mar  2 07:26:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 09:26:57 -0600
Subject: [Freeswitch-users] Orginate: getting status of call fail
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC17@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC04@sun.Noble.local>
	<191c3a030903020559s72d3e750ve4324ea614ee76ea@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C408BC17@sun.Noble.local>
Message-ID: <191c3a030903020726y455b786dka15206f2be5a7559@mail.gmail.com>

yes if you match the job uuid from bgapi to the SWITCH_EVENT_BACKGROUND_JOB
event, you would get the result in that event.


On Mon, Mar 2, 2009 at 8:49 AM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  That?s what I was wondering, however, won?t the response to the bagi (not
> the initial) give me the info on the call result?
>
>
>
> Regards
>
>
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* 02 March 2009 14:00
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Orginate: getting status of call fail
>
>
>
> The best way would be to add a few custom variables and add a secondary
> system that monitors the CDR data and uses the
> custom variables to identify what you want to do with the failed calls.
>
>
>  On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton <
> nik.middleton at noblesolutions.co.uk> wrote:
>
> Hi Guys,
>
>
>
> I?ve been running a test script written in lua which now works very well
> thanks to Anthony?s fix to stream file.
>
>
>
> Right now I?m using an event socket to initiate the call and passing the
> name of the script along with originate thus:
>
>
>
> $dialstring = "originate
> {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum
> '&lua(helloworld.lua )'";
>
> $result = $obj ->bgapi_command($dialstring);
>
>
>
> The script gets fired (it would appear) on answer.  However, if the number
> is invalid , timed out or was busy, I?m not sure the script gets executed or
> am I wrong?
>
>
>
> I want to be able to fire an event back on what happed to the call in the
> event that it failed for whatever reason.
>
>
>
> I know I can simply call the originate and pass the number as an argument
> and execute the dial within the script but I?m led to believe that?s not
> very efficient, or am I completely wrong?
>
>
>
> Looking for the most FS friendly way here
>
>
>
> Regards,
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From a.playful.idiot at gmail.com  Sun Mar  1 23:43:34 2009
From: a.playful.idiot at gmail.com (Aplayful Idiot)
Date: Sun, 1 Mar 2009 23:43:34 -0800
Subject: [Freeswitch-users] First time test set up FreeSwitch and
	SPA3102/SPA3000
Message-ID: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com>

I have no background in telephony but probably need to use a PBX.

FreeSwitch was recommended by a casual contact so I would like to start
first by setting up a small test.

I have a SPA3102 attached to the box running FS and to a ordinary phone
line. I registered SPA in conf/directory/default/line1.xml and it works ok
but I can't get caller id numbers from incoming calls. All FS sees is
"line1" which is found in file line1.xml as <variable
name="effective_caller_id_number" value="line1"/>.

Looking back over the FS wiki, I'm now wondering if the SPA should of been
set-up as a gateway but reading it is confusing at least to me. Sometimes I
think the analogue-line-SPA-FS is like a softphone which is registered to an
extension numbered xml file in conf/directory/default/ but then issues like
not getting outside incoming caller id's makes me think I've got this all
wrong.

Can someone help me out with this?

api
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From a.playful.idiot at gmail.com  Mon Mar  2 07:41:31 2009
From: a.playful.idiot at gmail.com (Aplayful Idiot)
Date: Mon, 2 Mar 2009 07:41:31 -0800
Subject: [Freeswitch-users] First time setting up FreeSwitch and SPA3102 /
	SPA3000
Message-ID: <9ed22e920903020741o2ff66970h35d3182655b2c7ba@mail.gmail.com>

Hi.

I have little background in telephony and need to use a PBX but would like
to start first with a small test set-up.

I have a SPA3102 attached to the box running FS and to a ordinary phone
line.

I registered SPA in conf/directory/default/line1.xml and this works to a
point but I can't get caller id numbers from incoming calls. All FS sees is
"line1" which is found in file line1.xml as <variable
name="effective_caller_id_number" value="line1"/>.

Looking back over the FS wiki, I'm now wondering if the SPA was registered
or set-up in FS correctly but reading the documentation is confusing me a
bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone
which is registered to an extension numbered xml file in
conf/directory/default/ but then issues like not getting outside incoming
caller id's makes me think I've got this all wrong.

Can someone help me out with this?

Thanks.
api
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From mike at jerris.com  Mon Mar  2 08:14:52 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 2 Mar 2009 11:14:52 -0500
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
In-Reply-To: <49AB8562.4050806@laposte.net>
References: <49AB8562.4050806@laposte.net>
Message-ID: <E437058E-E930-4060-B71C-80192165614D@jerris.com>

Could you please post this to jira along with a thread apply all bt of  
a core file taken from the process with the stuck sessions.

Mike

On Mar 2, 2009, at 2:06 AM, rod wrote:

> Hi All,
>
> I ran some longer tests with FS 1.0.3 acting as an SBC.
> The test machine has the following specs:
>    - Intel Quad Core Q9550
>    - 8GB RAM (far too much from what I saw)
>
> After 3 days running SIPP with 750 simultaneous calls (1500  
> channels) at
> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.
>
> In the CLI, using status command I got this:
>
> freeswitch at internal> status
> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds,
> 607 microseconds
> 15817560 session(s) since startup
> 22 session(s) 0/500
>
> But when I use "show channels" or "show calls", I see nothing. So I'm
> wondering where are these 22 sessions ?
>
> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs.
>
> Successful call -->  5271434
> Failed call ---> 1554      (less than 0.03%)
>
> regards,
> rod.
>
>
>
> complete SIPP summary:
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>  Call-rate(length)     Port   Total-time  Total-calls  Remote-host
> 50.0(35000 ms)/1.000s   5060  254259.42 s      5273022
> 10.10.10.254:5060(UDP)
>
>  0 new calls during 0.856 s period      7 ms scheduler resolution
>  0 calls (limit 750)                    Peak was 750 calls, after 15 s
>  0 Running, 34 Paused, 0 Woken up
>  15544 out-of-call msg (discarded)
>  1 open sockets
>  9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate
> (kB/s)
>  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate  
> (kB/s)
>
>                                 Messages  Retrans   Timeout
> Unexpected-Msg
>      INVITE ---------->         5273022   0         0
>         100 <----------         5273022   0                   1554
>         180 <----------         0         0                   0
>         183 <----------         0         0                   0
>         200 <----------  E-RTD1 5271434   0                   0
>         ACK ---------->         5271434   0
>       Pause [    35.0s]         5271434                       0
>         BYE ---------->         5271434   0         0
>         200 <----------         5271434   0                   0
>
> ------------------------------ Test Terminated
> --------------------------------
>
>
> ----------------------------- Statistics Screen ------- [1-9]: Change
> Screen --
>  Start Time             | 2009-02-27
> 09:11:31
>  Last Reset Time        | 2009-03-02
> 07:49:10
>  Current Time           | 2009-03-02
> 07:49:11
> -------------------------+--------------------------- 
> +--------------------------
>  Counter Name           | Periodic value            | Cumulative value
> -------------------------+--------------------------- 
> +--------------------------
>  Elapsed Time           | 00:00:00:857              |
> 70:37:39:429
>  Call Rate              |    0.000 cps              |   20.739
> cps
> -------------------------+--------------------------- 
> +--------------------------
>  Incoming call created  |        0                  |
> 0
>  OutGoing call created  |        0                  |
> 5273022
>  Total Call created     |                           |
> 5273022
>  Current Call           |       34
> |
> -------------------------+--------------------------- 
> +--------------------------
>  Successful call        |        0                  |
> 5271434
>  Failed call            |        0                  |
> 1554
> -------------------------+--------------------------- 
> +--------------------------
>  Response Time 1        | 00:00:00:000              |
> 00:00:00:240
>  Call Length            | 38:32:13:386              |
> 00:00:36:131
> ------------------------------ Test Terminated
> --------------------------------



From mike at jerris.com  Mon Mar  2 08:18:42 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 2 Mar 2009 11:18:42 -0500
Subject: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?
In-Reply-To: <7.0.1.0.2.20090302124612.028657a8@free.fr>
References: <mailman.6058.1235986943.18013.freeswitch-users@lists.freeswitch.org>
	<7.0.1.0.2.20090302124612.028657a8@free.fr>
Message-ID: <485E559F-4D98-48AB-8CF0-EF3FD50DBB5F@jerris.com>

I think any issues we have are related to pri, the analog doesn't seem  
to generate any major bug reports.

Mike

On Mar 2, 2009, at 6:47 AM, Fred wrote:

> Thanks guys for the feedback. So, the OpenZap driver isn't ready for
> production yet?
>



From Prometheus001 at gmx.net  Mon Mar  2 08:19:32 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 02 Mar 2009 17:19:32 +0100
Subject: [Freeswitch-users] hide caller_id_number,
 transfer and then  change caller_id_number
In-Reply-To: <191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>
References: <49AB004C.6090604@gmx.net>
	<191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>
Message-ID: <49AC0714.8000006@gmx.net>

Hello Anthony,

I do this when I orginate the call. This way we suppress the cid when we
call party A and transfer A to an internal extension (our callback
application).
But now comes the part that does not work:
After A enters the target number B (via DTMF), we set the cid variables
via uuid_setvar and then transfer A via uuid_transfer to party B.
However uuid_setvar does not work in that case.

BUT: If we do the same scenario and do not suppress the cid in the
originate part, then uuid_setvar works correctly and sets the cid_number.

Best regards
Peter

Anthony Minessale schrieb:
> put origination_caller_id_number in the dial string of any call and
> you can set the caller id individually for that leg
>
> {origination_caller_id_number=1234}<any normal dial string>
>
>
> On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     I have the following problem while providing callback (mod_eventsocket
>     is used):
>     1) I want to call a certain destination number A with a suppressed
>     caller_id_number (this works fine with some vars in the
>     origination string)
>     2) The destination number A picks up the phone and enters a target
>     number B by DTMF
>     3) freeswitch then forwards the call to target number B by DTMF and I
>     want to show the number A. I do this with uuid_setvar. The problem is
>     that it still shows unknown.
>     This is all with SIP.
>
>     uuid_setvar however worked if I did not set the caller_id_number to
>     unknown. Per default this is then "00000000000" and can then be
>     changed
>     with uuid_setvar to the number of A.
>     But if I set caller_id_number to unknown I can no longer change it
>     to A.
>
>     Any hint?
>
>     Best regards
>     Peter
>
>
>
>
>
>
>     _______________________________________________
>     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/
>
> 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>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 woof at nortel.com  Mon Mar  2 09:16:33 2009
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 02 Mar 2009 12:16:33 -0500
Subject: [Freeswitch-users] Detecting the origin of voice activity using
 VAD
In-Reply-To: <29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
Message-ID: <op.up6cxvrrssieo1@woof3.fuzzy>

Woof!

On Sun, 01 Mar 2009 21:28:18 -0500, Brian West <brian at freeswitch.org> wrote:

> NO.  You want something that people THINK exists and works well...
> Reliable human/voice detection doesn't exist in ANY form.

I beg to differ.  See http://www.freepatentsonline.com/5521967.html for one way to do it.  It works rather well and can quickly descriminate between voice and tone.  I've no idea who owns that patent now (not me, for sure).

There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented.  If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample.  So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone.  Voice will have dramatic power level differences over that same period.  This works very well in today's telephony environment, where tones are computer generated.  In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well.

--Woof!


From msc at freeswitch.org  Mon Mar  2 10:05:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Mar 2009 10:05:43 -0800
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49A7F393.6080406@ewetel.de>
References: <49A7F393.6080406@ewetel.de>
Message-ID: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>

On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:
> Hello,
>
> I play around with record_session and would like to have caller and
> callee separated on left and right channel. I found record_stereo is
> used for this. Unfortunately it doesn't work. A and B leg are still
> mixed. Additionally I found that B leg is significant louder than A leg,
> but both legs were local extensions.
>
Just to confirm - you are trying to record each leg of the call into a
separate file? In other words, one call creates two separate audio
recordings?

-MC


From anthony.minessale at gmail.com  Mon Mar  2 11:48:26 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 13:48:26 -0600
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <op.up6cxvrrssieo1@woof3.fuzzy>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<op.up6cxvrrssieo1@woof3.fuzzy>
Message-ID: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com>

i think that's what mod_vmd does

On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer <woof at nortel.com> wrote:

> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West <brian at freeswitch.org>
> wrote:
>
> > NO.  You want something that people THINK exists and works well...
> > Reliable human/voice detection doesn't exist in ANY form.
>
> I beg to differ.  See http://www.freepatentsonline.com/5521967.html for
> one way to do it.  It works rather well and can quickly descriminate between
> voice and tone.  I've no idea who owns that patent now (not me, for sure).
>
> There is a simpler, less reliable way of differentiating voice from tone,
> that as far as I know isn't patented.  If you compare the RMS power levels
> of sequential 40 mS periods, call progress tones will have very consistent
> power levels from sample to sample.  So if 5 or more 40 mS periods have
> about the same power measurement (within say, 2%), it's a tone.  Voice will
> have dramatic power level differences over that same period.  This works
> very well in today's telephony environment, where tones are computer
> generated.  In the old days when ringback tone was generated off the audio
> hum from the 20 Hz ring voltage generator...not so well.
>
> --Woof!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Mar  2 11:52:16 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 13:52:16 -0600
Subject: [Freeswitch-users] hide caller_id_number,
	transfer and then 	change caller_id_number
In-Reply-To: <49AC0714.8000006@gmx.net>
References: <49AB004C.6090604@gmx.net>
	<191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>
	<49AC0714.8000006@gmx.net>
Message-ID: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com>

origination_caller_id number is not ok as a variable unless its in {} as
part of the dial string
it's an exception that is parsed before the channel is even created.

I think you are drawing the wrong conclusion about what works and doesn't
work.
If you can produce a dial string that contains
{origination_caller_id_number=x} you will always be able to set it.

I assume you are using a recent version of FS as we did have a small bug
with this variable a few weeks ago.


On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Anthony,
>
> I do this when I orginate the call. This way we suppress the cid when we
> call party A and transfer A to an internal extension (our callback
> application).
> But now comes the part that does not work:
> After A enters the target number B (via DTMF), we set the cid variables
> via uuid_setvar and then transfer A via uuid_transfer to party B.
> However uuid_setvar does not work in that case.
>
> BUT: If we do the same scenario and do not suppress the cid in the
> originate part, then uuid_setvar works correctly and sets the cid_number.
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > put origination_caller_id_number in the dial string of any call and
> > you can set the caller id individually for that leg
> >
> > {origination_caller_id_number=1234}<any normal dial string>
> >
> >
> > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Hello,
> >
> >     I have the following problem while providing callback
> (mod_eventsocket
> >     is used):
> >     1) I want to call a certain destination number A with a suppressed
> >     caller_id_number (this works fine with some vars in the
> >     origination string)
> >     2) The destination number A picks up the phone and enters a target
> >     number B by DTMF
> >     3) freeswitch then forwards the call to target number B by DTMF and I
> >     want to show the number A. I do this with uuid_setvar. The problem is
> >     that it still shows unknown.
> >     This is all with SIP.
> >
> >     uuid_setvar however worked if I did not set the caller_id_number to
> >     unknown. Per default this is then "00000000000" and can then be
> >     changed
> >     with uuid_setvar to the number of A.
> >     But if I set caller_id_number to unknown I can no longer change it
> >     to A.
> >
> >     Any hint?
> >
> >     Best regards
> >     Peter
> >
> >
> >
> >
> >
> >
> >     _______________________________________________
> >     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/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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From msc at freeswitch.org  Mon Mar  2 12:03:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Mar 2009 12:03:52 -0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<op.up6cxvrrssieo1@woof3.fuzzy>
	<191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com>
Message-ID: <87f2f3b90903021203y3588ffa4nc3ccd280117e0129@mail.gmail.com>

On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> i think that's what mod_vmd does
>
I think that's right. It just does the opposite - instead of looking
for differing power levels it looks for the same power level. In other
words it tries to detect distinctly non-human sound. I'll bet you
could futz with that code and have it fire off events when it detects
what it believes is human speech.

-MC


From Prometheus001 at gmx.net  Mon Mar  2 12:31:47 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 02 Mar 2009 21:31:47 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49AB0BCD.8030108@gmx.net>
References: <49A92BAE.4090907@gmx.net>	<F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>
	<49AB0BCD.8030108@gmx.net>
Message-ID: <49AC4233.6060506@gmx.net>

Some more info:
the system I am working on is a copy (dd copy) of a system where the
pizza demo works on.
The only thing I changed was to update to the current freeswitch trunk
12293 (it was 10003 before).

Do I need to update the model? I did a make in the model directory, but
no change.

Best regards
Peter

Peter P GMX schrieb:
> Hello Brian,
>
> thanks for the info. I am a step further, but it cannot load the grammar
> files.
> I am sending through event_socket:
>
> SendMsg
> call-command: execute
> execute-app-name: detect_speech
> execute-app-arg: pocketsphinx yes no
>
> However I get the message (also when I am using Pizza demo):
> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
> sofia/internal/1000 at sip2.server.com Command Execute
> detect_speech(pocketsphinx yes no)
> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
> pocketsphinx_asr_load_grammar() Can't open language model
> /usr/local/freeswitch/grammar/model/communicator.
> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
> switch_ivr_detect_speech() Error loading Grammar
> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
> pocketsphinx_asr_close() Port Closed.
>
> However the grammar files are there:
> root at sip2:/usr/local/freeswitch/grammar/model/communicator#
> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
> total 12752
> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances
>
>
> Any hint?
>
> Best regards
> Peter
>
> Brian West schrieb:
>   
>> You can accomplish this .... here is an example using ESL in perl
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344
>>
>> /b
>>
>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:
>>
>>   
>>     
>>> Or back to the basics: Is it possible to use pocketsphinx through  
>>> event
>>> socket?
>>>     
>>>       
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>   
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 sergio.alecha at gmail.com  Mon Mar  2 12:18:06 2009
From: sergio.alecha at gmail.com (Sergio Alecha)
Date: Mon, 2 Mar 2009 17:18:06 -0300
Subject: [Freeswitch-users] Howto config early dial
Message-ID: <47f9a0940903021218l1a6f7ea1l8c596ddd3514e86c@mail.gmail.com>

In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use
"early dial" Grandstream telephones. How do Freeswitch in?
thank you very much.
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From Prometheus001 at gmx.net  Mon Mar  2 13:58:35 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 02 Mar 2009 22:58:35 +0100
Subject: [Freeswitch-users] hide caller_id_number,
 transfer and then  change caller_id_number
In-Reply-To: <191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com>
References: <49AB004C.6090604@gmx.net>	<191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>	<49AC0714.8000006@gmx.net>
	<191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com>
Message-ID: <49AC568B.7020504@gmx.net>

Hello Anthony,

sorry for being tenacious but in some cases it works in a way we need it:
If I a am not suppressing the cid numer when calling A, the following
scenario works:

    * A receives a Call (originate) with CID '0000000000' (default from
      switch_caller.c)
    * A dials some digits via DTMF, the app set the cid variables via
      uuid_setvar and uuid_transfers the call to B. B receives a call
      with the right cid set. 

Maybe I simply modify the default cid '0000000000'  to a different value
in switch_caller.c? Is there a special reason why this is '0000000000'?

I am using trunk version 12293.

Best regards
Peter

Anthony Minessale schrieb:
> origination_caller_id number is not ok as a variable unless its in {}
> as part of the dial string
> it's an exception that is parsed before the channel is even created.
>
> I think you are drawing the wrong conclusion about what works and
> doesn't work.
> If you can produce a dial string that contains
> {origination_caller_id_number=x} you will always be able to set it.
>
> I assume you are using a recent version of FS as we did have a small
> bug with this variable a few weeks ago.
>
>
> On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello Anthony,
>
>     I do this when I orginate the call. This way we suppress the cid
>     when we
>     call party A and transfer A to an internal extension (our callback
>     application).
>     But now comes the part that does not work:
>     After A enters the target number B (via DTMF), we set the cid
>     variables
>     via uuid_setvar and then transfer A via uuid_transfer to party B.
>     However uuid_setvar does not work in that case.
>
>     BUT: If we do the same scenario and do not suppress the cid in the
>     originate part, then uuid_setvar works correctly and sets the
>     cid_number.
>
>     Best regards
>     Peter
>
>     Anthony Minessale schrieb:
>     > put origination_caller_id_number in the dial string of any call and
>     > you can set the caller id individually for that leg
>     >
>     > {origination_caller_id_number=1234}<any normal dial string>
>     >
>     >
>     > On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX
>     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
>     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
>     wrote:
>     >
>     >     Hello,
>     >
>     >     I have the following problem while providing callback
>     (mod_eventsocket
>     >     is used):
>     >     1) I want to call a certain destination number A with a
>     suppressed
>     >     caller_id_number (this works fine with some vars in the
>     >     origination string)
>     >     2) The destination number A picks up the phone and enters a
>     target
>     >     number B by DTMF
>     >     3) freeswitch then forwards the call to target number B by
>     DTMF and I
>     >     want to show the number A. I do this with uuid_setvar. The
>     problem is
>     >     that it still shows unknown.
>     >     This is all with SIP.
>     >
>     >     uuid_setvar however worked if I did not set the
>     caller_id_number to
>     >     unknown. Per default this is then "00000000000" and can then be
>     >     changed
>     >     with uuid_setvar to the number of A.
>     >     But if I set caller_id_number to unknown I can no longer
>     change it
>     >     to A.
>     >
>     >     Any hint?
>     >
>     >     Best regards
>     >     Peter
>     >
>     >
>     >
>     >
>     >
>     >
>     >     _______________________________________________
>     >     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
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > iax:guest at conference.freeswitch.org/888
>     <http://iax:guest at conference.freeswitch.org/888>
>     > <http://iax:guest at conference.freeswitch.org/888>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:213-799-1400
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > 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
>     >
>
>     _______________________________________________
>     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/
>
> 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>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Mar  2 14:08:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 2 Mar 2009 14:08:53 -0800
Subject: [Freeswitch-users] hide caller_id_number,
	transfer and then 	change caller_id_number
In-Reply-To: <49AC568B.7020504@gmx.net>
References: <49AB004C.6090604@gmx.net>
	<191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>
	<49AC0714.8000006@gmx.net>
	<191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com>
	<49AC568B.7020504@gmx.net>
Message-ID: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com>

On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Anthony,
>
> sorry for being tenacious but in some cases it works in a way we need it:
> If I a am not suppressing the cid numer when calling A, the following
> scenario works:
>
> ? ?* A receives a Call (originate) with CID '0000000000' (default from
> ? ? ?switch_caller.c)
> ? ?* A dials some digits via DTMF, the app set the cid variables via
> ? ? ?uuid_setvar and uuid_transfers the call to B. B receives a call
> ? ? ?with the right cid set.
>
> Maybe I simply modify the default cid '0000000000' ?to a different value
> in switch_caller.c? Is there a special reason why this is '0000000000'?
>

Check vars.xml to confirm that you have actually set a default caller
ID. Most likely you'll still have the default caller id number set to
all zeroes, which is the default.

-MC

> I am using trunk version 12293.
>
> Best regards
> Peter
>


From anthony.minessale at gmail.com  Mon Mar  2 14:22:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 2 Mar 2009 16:22:25 -0600
Subject: [Freeswitch-users] hide caller_id_number,
	transfer and then 	change caller_id_number
In-Reply-To: <87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com>
References: <49AB004C.6090604@gmx.net>
	<191c3a030903020556h36b7aa00j5b6dd978b2211aa@mail.gmail.com>
	<49AC0714.8000006@gmx.net>
	<191c3a030903021152y6d7b0ab9wede51d39671aeda2@mail.gmail.com>
	<49AC568B.7020504@gmx.net>
	<87f2f3b90903021408g62ba6a4eo925b70beb1a562a5@mail.gmail.com>
Message-ID: <191c3a030903021422i385ab5c0h781993c65d6d27b4@mail.gmail.com>

Since you did not describe the exact way you are doing it with enough detail
or any trace I can't begin to tell you
what your problem is.  you did not even mention what variable you are using
or show examples.

All I can do is tell you again that if you set the
origination_caller_id_number in the dial string it will
be the most likely to work for you.




On Mon, Mar 2, 2009 at 4:08 PM, Michael Collins <msc at freeswitch.org> wrote:

> On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> > Hello Anthony,
> >
> > sorry for being tenacious but in some cases it works in a way we need it:
> > If I a am not suppressing the cid numer when calling A, the following
> > scenario works:
> >
> >    * A receives a Call (originate) with CID '0000000000' (default from
> >      switch_caller.c)
> >    * A dials some digits via DTMF, the app set the cid variables via
> >      uuid_setvar and uuid_transfers the call to B. B receives a call
> >      with the right cid set.
> >
> > Maybe I simply modify the default cid '0000000000'  to a different value
> > in switch_caller.c? Is there a special reason why this is '0000000000'?
> >
>
> Check vars.xml to confirm that you have actually set a default caller
> ID. Most likely you'll still have the default caller id number set to
> all zeroes, which is the default.
>
> -MC
>
> > I am using trunk version 12293.
> >
> > Best regards
> > Peter
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From freeswitch at servercorps.com  Mon Mar  2 14:43:04 2009
From: freeswitch at servercorps.com (Addison Martin)
Date: Mon, 2 Mar 2009 16:43:04 -0600
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49AC4233.6060506@gmx.net>
References: <49A92BAE.4090907@gmx.net>
	<F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>
	<49AB0BCD.8030108@gmx.net> <49AC4233.6060506@gmx.net>
Message-ID: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com>

Peter,

You need the grammar files for the pizza demo:
http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo
has lonks to premade fles for everyhting to get the pizza demo working
with pocketshinx.  Those to not come with the source code when you
update from SVN.

Nik



On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Some more info:
> the system I am working on is a copy (dd copy) of a system where the
> pizza demo works on.
> The only thing I changed was to update to the current freeswitch trunk
> 12293 (it was 10003 before).
>
> Do I need to update the model? I did a make in the model directory, but
> no change.
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
>> Hello Brian,
>>
>> thanks for the info. I am a step further, but it cannot load the grammar
>> files.
>> I am sending through event_socket:
>>
>> SendMsg
>> call-command: execute
>> execute-app-name: detect_speech
>> execute-app-arg: pocketsphinx yes no
>>
>> However I get the message (also when I am using Pizza demo):
>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
>> sofia/internal/1000 at sip2.server.com Command Execute
>> detect_speech(pocketsphinx yes no)
>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
>> pocketsphinx_asr_load_grammar() Can't open language model
>> /usr/local/freeswitch/grammar/model/communicator.
>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
>> switch_ivr_detect_speech() Error loading Grammar
>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
>> pocketsphinx_asr_close() Port Closed.
>>
>> However the grammar files are there:
>> root at sip2:/usr/local/freeswitch/grammar/model/communicator#
>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
>> total 12752
>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances
>>
>>
>> Any hint?
>>
>> Best regards
>> Peter
>>
>> Brian West schrieb:
>>
>>> You can accomplish this .... here is an example using ESL in perl
>>>
>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344
>>>
>>> /b
>>>
>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:
>>>
>>>
>>>
>>>> Or back to the basics: Is it possible to use pocketsphinx through
>>>> event
>>>> socket?
>>>>
>>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Prometheus001 at gmx.net  Mon Mar  2 15:42:28 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 03 Mar 2009 00:42:28 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com>
References: <49A92BAE.4090907@gmx.net>	<F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>	<49AB0BCD.8030108@gmx.net>
	<49AC4233.6060506@gmx.net>
	<92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com>
Message-ID: <49AC6EE4.9080509@gmx.net>

Thanks Addison.
The Pizza files are there (as mentionned is it a copy of an already
working system).
In fact freeswitch is complaning about
/usr/local/freeswitch/grammar/model/communicator which he cannot load

So somehow freeswitch is not willing to open the files, but I have no
clue why. So any hints are welcome.

Best regards
Peter


Addison Martin schrieb:
> Peter,
>
> You need the grammar files for the pizza demo:
> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo
> has lonks to premade fles for everyhting to get the pizza demo working
> with pocketshinx.  Those to not come with the source code when you
> update from SVN.
>
> Nik
>
>
>
> On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Some more info:
>> the system I am working on is a copy (dd copy) of a system where the
>> pizza demo works on.
>> The only thing I changed was to update to the current freeswitch trunk
>> 12293 (it was 10003 before).
>>
>> Do I need to update the model? I did a make in the model directory, but
>> no change.
>>
>> Best regards
>> Peter
>>
>> Peter P GMX schrieb:
>>     
>>> Hello Brian,
>>>
>>> thanks for the info. I am a step further, but it cannot load the grammar
>>> files.
>>> I am sending through event_socket:
>>>
>>> SendMsg
>>> call-command: execute
>>> execute-app-name: detect_speech
>>> execute-app-arg: pocketsphinx yes no
>>>
>>> However I get the message (also when I am using Pizza demo):
>>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
>>> sofia/internal/1000 at sip2.server.com Command Execute
>>> detect_speech(pocketsphinx yes no)
>>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
>>> pocketsphinx_asr_load_grammar() Can't open language model
>>> /usr/local/freeswitch/grammar/model/communicator.
>>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
>>> switch_ivr_detect_speech() Error loading Grammar
>>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
>>> pocketsphinx_asr_close() Port Closed.
>>>
>>> However the grammar files are there:
>>> root at sip2:/usr/local/freeswitch/grammar/model/communicator#
>>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
>>> total 12752
>>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
>>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
>>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
>>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
>>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
>>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
>>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
>>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
>>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
>>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances
>>>
>>>
>>> Any hint?
>>>
>>> Best regards
>>> Peter
>>>
>>> Brian West schrieb:
>>>
>>>       
>>>> You can accomplish this .... here is an example using ESL in perl
>>>>
>>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344
>>>>
>>>> /b
>>>>
>>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:
>>>>
>>>>
>>>>
>>>>         
>>>>> Or back to the basics: Is it possible to use pocketsphinx through
>>>>> event
>>>>> socket?
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Mon Mar  2 16:08:27 2009
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 03 Mar 2009 08:08:27 +0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
 VAD
In-Reply-To: <op.up6cxvrrssieo1@woof3.fuzzy>
References: <49AACD71.5080103@teldio.com>	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<op.up6cxvrrssieo1@woof3.fuzzy>
Message-ID: <49AC74FB.3030903@coppice.org>

Andy Spitzer wrote:
> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West <brian at freeswitch.org> wrote:
>
>   
>> NO.  You want something that people THINK exists and works well...
>> Reliable human/voice detection doesn't exist in ANY form.
>>     
>
> I beg to differ.  See http://www.freepatentsonline.com/5521967.html for one way to do it.  It works rather well and can quickly descriminate between voice and tone.  I've no idea who owns that patent now (not me, for sure).
>   
Since when did a patent mean a problem is solved? For things like speech 
recognition you can achieve pretty high accuracy in voice detection, but 
in that case you can delay the audio and make decisions that span the 
start of the speech burst. For most telephony purposes you need to make 
a decision on the very first frame of speech, as you can't afford to add 
latency. That turns it into a tough problem. Something like the VAD in 
G.729 is about the best people can currently do, but its far from perfect.
> There is a simpler, less reliable way of differentiating voice from tone, that as far as I know isn't patented.  If you compare the RMS power levels of sequential 40 mS periods, call progress tones will have very consistent power levels from sample to sample.  So if 5 or more 40 mS periods have about the same power measurement (within say, 2%), it's a tone.  Voice will have dramatic power level differences over that same period.  This works very well in today's telephony environment, where tones are computer generated.  In the old days when ringback tone was generated off the audio hum from the 20 Hz ring voltage generator...not so well.
>   
That is *not* VAD. What you describe just says "is its energy steady". I 
will trigger on music, background noise and maybe even some of the fast 
pulsed tone signals. A proper VAD won't.

Regards,
Steve



From steveu at coppice.org  Mon Mar  2 16:10:50 2009
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 03 Mar 2009 08:10:50 +0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
 VAD
In-Reply-To: <191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com>
References: <49AACD71.5080103@teldio.com>	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>	<op.up6cxvrrssieo1@woof3.fuzzy>
	<191c3a030903021148t7ea7ba65j6a9b266e83a98bc3@mail.gmail.com>
Message-ID: <49AC758A.60304@coppice.org>

Hi,

mod_vmd is a bit more sophisticated than that. It looks for the signal 
being narrowband energy. However, mod_vmd isn't very reliable, as it 
takes a rather high SNR for its narrowband detector to work. So high 
that a lossy codec like G.711 can barely manage it.

Regards,
Steve

Anthony Minessale wrote:
> i think that's what mod_vmd does
>
> On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer <woof at nortel.com 
> <mailto:woof at nortel.com>> wrote:
>
>     Woof!
>
>     On Sun, 01 Mar 2009 21:28:18 -0500, Brian West
>     <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
>
>     > NO.  You want something that people THINK exists and works well...
>     > Reliable human/voice detection doesn't exist in ANY form.
>
>     I beg to differ.  See
>     http://www.freepatentsonline.com/5521967.html for one way to do
>     it.  It works rather well and can quickly descriminate between
>     voice and tone.  I've no idea who owns that patent now (not me,
>     for sure).
>
>     There is a simpler, less reliable way of differentiating voice
>     from tone, that as far as I know isn't patented.  If you compare
>     the RMS power levels of sequential 40 mS periods, call progress
>     tones will have very consistent power levels from sample to
>     sample.  So if 5 or more 40 mS periods have about the same power
>     measurement (within say, 2%), it's a tone.  Voice will have
>     dramatic power level differences over that same period.  This
>     works very well in today's telephony environment, where tones are
>     computer generated.  In the old days when ringback tone was
>     generated off the audio hum from the 20 Hz ring voltage
>     generator...not so well.
>
>     --Woof!
>
>     _______________________________________________
>     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
>



From steveu at coppice.org  Mon Mar  2 16:32:53 2009
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 03 Mar 2009 08:32:53 +0800
Subject: [Freeswitch-users] Detecting the origin of voice activity using
 VAD
In-Reply-To: <op.up6cxvrrssieo1@woof3.fuzzy>
References: <49AACD71.5080103@teldio.com>	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<op.up6cxvrrssieo1@woof3.fuzzy>
Message-ID: <49AC7AB5.5060505@coppice.org>

Andy Spitzer wrote:
> Woof!
>
> On Sun, 01 Mar 2009 21:28:18 -0500, Brian West <brian at freeswitch.org> wrote:
>
>   
>> NO.  You want something that people THINK exists and works well...
>> Reliable human/voice detection doesn't exist in ANY form.
>>     
>
> I beg to differ.  See http://www.freepatentsonline.com/5521967.html for one way to do it.  It works rather well and can quickly descriminate between voice and tone.  I've no idea who owns that patent now (not me, for sure).
>   
I just had a look through that patent. Its amazing. There is a lot of 
focussed descriptive text, but a patent only really consists of its 
claims. Those claims are astonishingly open-ended, and characterise what 
people had been doing for many years before it was filed - "Well we, er, 
make a call, we listen for some beeping, and we may hang up based on 
that". That is a really sick patent.

Steve



From mszlazak at aol.com  Mon Mar  2 16:49:09 2009
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Mon, 02 Mar 2009 19:49:09 -0500
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49AC6EE4.9080509@gmx.net>
References: <49AC6EE4.9080509@gmx.net>
Message-ID: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>




 I think you need to talk to Brian. 

Apparently this is a "new" pocketsphinx which works on a different format from those found in the pizza demo. 

Also, pocketsphinx crashes if it "hears" anything outside the grammar which apparently is a longstanding bug. Brian mentioned they are working on getting this fixed. 






I kept getting:


2009-02-25
19:49:32 [ERR] mod_pocketsphinx.c:140 pocketsphinx_asr_load_grammar()
Can't open dictionary
C:\Source\freeswitch-snapshot\Debug\grammar\default.dic.
2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed.

The suggestion was to "Just copy the cmudict.0.6d to
default.dic, not sure how well it will perform on windows.. if it does
badly you can slim the dictionary down to words you know you'll be
using."



https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d


That gave me more problems so I'm waiting for the fix.

Mark.







 





-----Original Message-----


From: Peter P GMX <Prometheus001 at gmx.net>


To: freeswitch-users at lists.freeswitch.org


Sent: Mon, 2 Mar 2009 3:42 pm


Subject: Re: [Freeswitch-users] pocketsphinx and event socket
















Thanks Addison.



The Pizza files are there (as mentionned is it a copy of an already



working system).



In fact freeswitch is complaning about



/usr/local/freeswitch/grammar/model/communicator which he cannot load







So somehow freeswitch is not willing to open the files, but I have no



clue why. So any hints are welcome.







Best regards



Peter











Addison Martin schrieb:



> Peter,



>



> You need the grammar files for the pizza demo:



> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo



> has lonks to premade fles for everyhting to get the pizza demo working



> with pocketshinx.  Those to not come with the source code when you



> update from SVN.



>



> Nik



>



>



>



> On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:



>   



>> Some more info:



>> the system I am working on is a copy (dd copy) of a system where the



>> pizza demo works on.



>> The only thing I changed was to update to the current freeswitch trunk



>> 12293 (it was 10003 before).



>>



>> Do I need to update the model? I did a make in the model directory, but



>> no change.



>>



>> Best regards



>> Peter



>>



>> Peter P GMX schrieb:



>>     



>>> Hello Brian,



>>>



>>> thanks for the info. I am a step further, but it cannot load the grammar



>>> files.



>>> I am sending through event_socket:



>>>



>>> SendMsg



>>> call-command: execute



>>> execute-app-name: detect_speech



>>> execute-app-arg: pocketsphinx yes no



>>>



>>> However I get the message (also when I am using Pizza demo):



>>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()



>>> sofia/internal/1000 at sip2.server.com Command Execute



>>> detect_speech(pocketsphinx yes no)



>>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145



>>> pocketsphinx_asr_load_grammar() Can't open language model



>>> /usr/local/freeswitch/grammar/model/communicator.



>>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041



>>> switch_ivr_detect_speech() Error loading Grammar



>>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219



>>> pocketsphinx_asr_close() Port Closed.



>>>



>>> However the grammar files are there:



>>> root at sip2:/usr/local/freeswitch/grammar/model/communicator#



>>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al



>>> total 12752



>>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .



>>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..



>>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING



>>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params



>>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef



>>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means



>>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict



>>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump



>>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices



>>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances



>>>



>>>



>>> Any hint?



>>>



>>> Best regards



>>> Peter



>>>



>>> Brian West schrieb:



>>>



>>>       



>>>> You can accomplish this .... here is an example using ESL in perl



>>>>



>>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344



>>>>



>>>> /b



>>>>



>>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:



>>>>



>>>>



>>>>



>>>>         



>>>>> Or back to the basics: Is it possible to use pocketsphinx through



>>>>> event



>>>>> socket?



>>>>>



>>>>>



>>>>>           



>>>> _______________________________________________



>>>> Freeswitch-users mailing list



>>>> Freeswitch-users at lists.freeswitch.org



>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



>>>> http://www.freeswitch.org



>>>>



>>>>



>>>>



>>>>         



>>> _______________________________________________



>>> Freeswitch-users mailing list



>>> Freeswitch-users at lists.freeswitch.org



>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



>>> http://www.freeswitch.org



>>>



>>>



>>>       



>> _______________________________________________



>> Freeswitch-users mailing list



>> Freeswitch-users at lists.freeswitch.org



>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



>> http://www.freeswitch.org



>>



>>     



>



> _______________________________________________



> Freeswitch-users mailing list



> Freeswitch-users at lists.freeswitch.org



> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users



> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users



> http://www.freeswitch.org



>



>   







_______________________________________________



Freeswitch-users mailing list



Freeswitch-users at 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 woof at nortel.com  Mon Mar  2 16:51:01 2009
From: woof at nortel.com (Andy Spitzer)
Date: Mon, 02 Mar 2009 19:51:01 -0500
Subject: [Freeswitch-users] Detecting the origin of voice activity using
 VAD
In-Reply-To: <49AC7AB5.5060505@coppice.org>
References: <49AACD71.5080103@teldio.com>
	<773ACF68-7B2C-481C-9328-6C42BEA968AA@freeswitch.org>
	<59ad9ca10903011820i77b98a7ah562c66008372177d@mail.gmail.com>
	<29C5E4C1-93AD-4E27-AAC4-D54F6FF7336F@freeswitch.org>
	<op.up6cxvrrssieo1@woof3.fuzzy> <49AC7AB5.5060505@coppice.org>
Message-ID: <op.up6xzbcbssieo1@woof3.fuzzy>

Woof!

On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood <steveu at coppice.org> wrote:

 I just had a look through that patent. Its amazing. There is a lot of
> focussed descriptive text, but a patent only really consists of its
> claims. Those claims are astonishingly open-ended, and characterise what
> people had been doing for many years before it was filed - "Well we, er,
> make a call, we listen for some beeping, and we may hang up based on
> that". That is a really sick patent.

Yep, I agree.  It was the ferping lawyers who kept "adding value" to try to broaden it.  What we (the inventors) wrote up was nice and clean.  It does have some new and novel technical approaches that we really did come up with...and could find no prior art for.  Then the lawyers got to it.  A true example of what's wrong with software patents these days.

--Woof!


From kawarod at laposte.net  Mon Mar  2 23:07:32 2009
From: kawarod at laposte.net (rod)
Date: Tue, 03 Mar 2009 11:07:32 +0400
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
In-Reply-To: <E437058E-E930-4060-B71C-80192165614D@jerris.com>
References: <49AB8562.4050806@laposte.net>
	<E437058E-E930-4060-B71C-80192165614D@jerris.com>
Message-ID: <49ACD734.7000700@laposte.net>

Hi Michael,

I checked on wiki, is the following the good way to go (sorry I'm not 
very familiar with your debugging tool).

$ gdb bin/freeswitch core.xxx

bt
bt full
thread apply all bt
thread apply all bt full 


If I understand well I have to rerun the tests, as I did not start FS 
using GDB.

regards,
rod




Michael Jerris wrote:
> Could you please post this to jira along with a thread apply all bt of  
> a core file taken from the process with the stuck sessions.
>
> Mike
>
> On Mar 2, 2009, at 2:06 AM, rod wrote:
>
>   
>> Hi All,
>>
>> I ran some longer tests with FS 1.0.3 acting as an SBC.
>> The test machine has the following specs:
>>    - Intel Quad Core Q9550
>>    - 8GB RAM (far too much from what I saw)
>>
>> After 3 days running SIPP with 750 simultaneous calls (1500  
>> channels) at
>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.
>>
>> In the CLI, using status command I got this:
>>
>> freeswitch at internal> status
>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds,
>> 607 microseconds
>> 15817560 session(s) since startup
>> 22 session(s) 0/500
>>
>> But when I use "show channels" or "show calls", I see nothing. So I'm
>> wondering where are these 22 sessions ?
>>
>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs.
>>
>> Successful call -->  5271434
>> Failed call ---> 1554      (less than 0.03%)
>>
>> regards,
>> rod.
>>
>>
>>
>> complete SIPP summary:
>>
>> ------------------------------ Scenario Screen -------- [1-9]: Change
>> Screen --
>>  Call-rate(length)     Port   Total-time  Total-calls  Remote-host
>> 50.0(35000 ms)/1.000s   5060  254259.42 s      5273022
>> 10.10.10.254:5060(UDP)
>>
>>  0 new calls during 0.856 s period      7 ms scheduler resolution
>>  0 calls (limit 750)                    Peak was 750 calls, after 15 s
>>  0 Running, 34 Paused, 0 Woken up
>>  15544 out-of-call msg (discarded)
>>  1 open sockets
>>  9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate
>> (kB/s)
>>  0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate  
>> (kB/s)
>>
>>                                 Messages  Retrans   Timeout
>> Unexpected-Msg
>>      INVITE ---------->         5273022   0         0
>>         100 <----------         5273022   0                   1554
>>         180 <----------         0         0                   0
>>         183 <----------         0         0                   0
>>         200 <----------  E-RTD1 5271434   0                   0
>>         ACK ---------->         5271434   0
>>       Pause [    35.0s]         5271434                       0
>>         BYE ---------->         5271434   0         0
>>         200 <----------         5271434   0                   0
>>
>> ------------------------------ Test Terminated
>> --------------------------------
>>
>>
>> ----------------------------- Statistics Screen ------- [1-9]: Change
>> Screen --
>>  Start Time             | 2009-02-27
>> 09:11:31
>>  Last Reset Time        | 2009-03-02
>> 07:49:10
>>  Current Time           | 2009-03-02
>> 07:49:11
>> -------------------------+--------------------------- 
>> +--------------------------
>>  Counter Name           | Periodic value            | Cumulative value
>> -------------------------+--------------------------- 
>> +--------------------------
>>  Elapsed Time           | 00:00:00:857              |
>> 70:37:39:429
>>  Call Rate              |    0.000 cps              |   20.739
>> cps
>> -------------------------+--------------------------- 
>> +--------------------------
>>  Incoming call created  |        0                  |
>> 0
>>  OutGoing call created  |        0                  |
>> 5273022
>>  Total Call created     |                           |
>> 5273022
>>  Current Call           |       34
>> |
>> -------------------------+--------------------------- 
>> +--------------------------
>>  Successful call        |        0                  |
>> 5271434
>>  Failed call            |        0                  |
>> 1554
>> -------------------------+--------------------------- 
>> +--------------------------
>>  Response Time 1        | 00:00:00:000              |
>> 00:00:00:240
>>  Call Length            | 38:32:13:386              |
>> 00:00:36:131
>> ------------------------------ Test Terminated
>> --------------------------------
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mrene_lists at avgs.ca  Mon Mar  2 23:56:01 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 3 Mar 2009 02:56:01 -0500
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
In-Reply-To: <49ACD734.7000700@laposte.net>
References: <49AB8562.4050806@laposte.net>
	<E437058E-E930-4060-B71C-80192165614D@jerris.com>
	<49ACD734.7000700@laposte.net>
Message-ID: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca>

Yes, you may also link (or copy) the .gdbinit file found in the  
support-d folder to your home directory.
This is going to enable some GDB macros written for FS.

Once thats done you can do the following commands and include them too:

list_sessions

hash_it_str_x session_manager.session_table switch_core_session_t  
channel->state


Its important to know that what you see in "show channels" and "show  
calls" is just a DB query to sqlite, Those commands will go directly  
in the core and list those sessions.

Math

On 3-Mar-09, at 2:07 AM, rod wrote:

> Hi Michael,
>
> I checked on wiki, is the following the good way to go (sorry I'm not
> very familiar with your debugging tool).
>
> $ gdb bin/freeswitch core.xxx
>
> bt
> bt full
> thread apply all bt
> thread apply all bt full
>
>
> If I understand well I have to rerun the tests, as I did not start FS
> using GDB.
>
> regards,
> rod
>
>
>
>
> Michael Jerris wrote:
>> Could you please post this to jira along with a thread apply all bt  
>> of
>> a core file taken from the process with the stuck sessions.
>>
>> Mike
>>
>> On Mar 2, 2009, at 2:06 AM, rod wrote:
>>
>>
>>> Hi All,
>>>
>>> I ran some longer tests with FS 1.0.3 acting as an SBC.
>>> The test machine has the following specs:
>>>   - Intel Quad Core Q9550
>>>   - 8GB RAM (far too much from what I saw)
>>>
>>> After 3 days running SIPP with 750 simultaneous calls (1500
>>> channels) at
>>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.
>>>
>>> In the CLI, using status command I got this:
>>>
>>> freeswitch at internal> status
>>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859  
>>> milliseconds,
>>> 607 microseconds
>>> 15817560 session(s) since startup
>>> 22 session(s) 0/500
>>>
>>> But when I use "show channels" or "show calls", I see nothing. So  
>>> I'm
>>> wondering where are these 22 sessions ?
>>>
>>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free  
>>> CPUs.
>>>
>>> Successful call -->  5271434
>>> Failed call ---> 1554      (less than 0.03%)
>>>
>>> regards,
>>> rod.
>>>
>>>
>>>
>>> complete SIPP summary:
>>>
>>> ------------------------------ Scenario Screen -------- [1-9]:  
>>> Change
>>> Screen --
>>> Call-rate(length)     Port   Total-time  Total-calls  Remote-host
>>> 50.0(35000 ms)/1.000s   5060  254259.42 s      5273022
>>> 10.10.10.254:5060(UDP)
>>>
>>> 0 new calls during 0.856 s period      7 ms scheduler resolution
>>> 0 calls (limit 750)                    Peak was 750 calls, after  
>>> 15 s
>>> 0 Running, 34 Paused, 0 Woken up
>>> 15544 out-of-call msg (discarded)
>>> 1 open sockets
>>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP  
>>> rate
>>> (kB/s)
>>> 0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate
>>> (kB/s)
>>>
>>>                                Messages  Retrans   Timeout
>>> Unexpected-Msg
>>>     INVITE ---------->         5273022   0         0
>>>        100 <----------         5273022   0                   1554
>>>        180 <----------         0         0                   0
>>>        183 <----------         0         0                   0
>>>        200 <----------  E-RTD1 5271434   0                   0
>>>        ACK ---------->         5271434   0
>>>      Pause [    35.0s]         5271434                       0
>>>        BYE ---------->         5271434   0         0
>>>        200 <----------         5271434   0                   0
>>>
>>> ------------------------------ Test Terminated
>>> --------------------------------
>>>
>>>
>>> ----------------------------- Statistics Screen ------- [1-9]:  
>>> Change
>>> Screen --
>>> Start Time             | 2009-02-27
>>> 09:11:31
>>> Last Reset Time        | 2009-03-02
>>> 07:49:10
>>> Current Time           | 2009-03-02
>>> 07:49:11
>>> -------------------------+---------------------------
>>> +--------------------------
>>> Counter Name           | Periodic value            | Cumulative  
>>> value
>>> -------------------------+---------------------------
>>> +--------------------------
>>> Elapsed Time           | 00:00:00:857              |
>>> 70:37:39:429
>>> Call Rate              |    0.000 cps              |   20.739
>>> cps
>>> -------------------------+---------------------------
>>> +--------------------------
>>> Incoming call created  |        0                  |
>>> 0
>>> OutGoing call created  |        0                  |
>>> 5273022
>>> Total Call created     |                           |
>>> 5273022
>>> Current Call           |       34
>>> |
>>> -------------------------+---------------------------
>>> +--------------------------
>>> Successful call        |        0                  |
>>> 5271434
>>> Failed call            |        0                  |
>>> 1554
>>> -------------------------+---------------------------
>>> +--------------------------
>>> Response Time 1        | 00:00:00:000              |
>>> 00:00:00:240
>>> Call Length            | 38:32:13:386              |
>>> 00:00:36:131
>>> ------------------------------ Test Terminated
>>> --------------------------------
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 helmut.kuper at ewetel.de  Tue Mar  3 00:16:02 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 03 Mar 2009 09:16:02 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>
References: <49A7F393.6080406@ewetel.de>
	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>
Message-ID: <49ACE742.5090809@ewetel.de>

Hi Mike,

no, I want just one file in stereo, where left channel is caller and
right channel is callee.

regards
Helmut

On 02.03.2009 19:05, Michael Collins wrote:
>
> Just to confirm - you are trying to record each leg of the call into a
> separate file? In other words, one call creates two separate audio
> recordings?
>
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Prometheus001 at gmx.net  Tue Mar  3 03:07:39 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Tue, 03 Mar 2009 12:07:39 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>
References: <49AC6EE4.9080509@gmx.net>
	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>
Message-ID: <49AD0F7B.7000802@gmx.net>

Thanks Mark,

I now switched back to rev. 10003 and the Pizza works again.

Best regards
Petere

mszlazak at aol.com schrieb:
> I think you need to talk to Brian.
>
> Apparently this is a "new" pocketsphinx which works on a different
> format from those found in the pizza demo.
>
> Also, pocketsphinx crashes if it "hears" anything outside the grammar
> which apparently is a longstanding bug. Brian mentioned they are
> working on getting this fixed.
>
> I kept getting:
>
> 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140
> pocketsphinx_asr_load_grammar() Can't open dictionary
> C:\Source\freeswitch-snapshot\Debug\grammar\default.dic.
> 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219
> pocketsphinx_asr_close() Port Closed.
>
> The suggestion was to "Just copy the cmudict.0.6d to default.dic, not
> sure how well it will perform on windows.. if it does badly you can
> slim the dictionary down to words you know you'll be using."
>
> https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d
>
> That gave me more problems so I'm waiting for the fix.
>
> Mark.
>
>
>
> -----Original Message-----
> From: Peter P GMX <Prometheus001 at gmx.net>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Mon, 2 Mar 2009 3:42 pm
> Subject: Re: [Freeswitch-users] pocketsphinx and event socket
>
> Thanks Addison.
>
>
>
>
>
>
>
> The Pizza files are there (as mentionned is it a copy of an already
>
>
>
>
>
>
>
> working system).
>
>
>
>
>
>
>
> In fact freeswitch is complaning about
>
>
>
>
>
>
>
> /usr/local/freeswitch/grammar/model/communicator which he cannot load
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> So somehow freeswitch is not willing to open the files, but I have no
>
>
>
>
>
>
>
> clue why. So any hints are welcome.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Best regards
>
>
>
>
>
>
>
> Peter
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Addison Martin schrieb:
>
>
>
>
>
>
>
> > Peter,
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> > You need the grammar files for the pizza demo:
>
>
>
>
>
>
>
> > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo
>
>
>
>
>
>
>
> > has lonks to premade fles for everyhting to get the pizza demo working
>
>
>
>
>
>
>
> > with pocketshinx.  Those to not come with the source code when you
>
>
>
>
>
>
>
> > update from SVN.
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> > Nik
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> > On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>> wrote:
>
>
>
>
>
>
>
> >   
>
>
>
>
>
>
>
> >> Some more info:
>
>
>
>
>
>
>
> >> the system I am working on is a copy (dd copy) of a system where the
>
>
>
>
>
>
>
> >> pizza demo works on.
>
>
>
>
>
>
>
> >> The only thing I changed was to update to the current freeswitch trunk
>
>
>
>
>
>
>
> >> 12293 (it was 10003 before).
>
>
>
>
>
>
>
> >>
>
>
>
>
>
>
>
> >> Do I need to update the model? I did a make in the model directory, but
>
>
>
>
>
>
>
> >> no change.
>
>
>
>
>
>
>
> >>
>
>
>
>
>
>
>
> >> Best regards
>
>
>
>
>
>
>
> >> Peter
>
>
>
>
>
>
>
> >>
>
>
>
>
>
>
>
> >> Peter P GMX schrieb:
>
>
>
>
>
>
>
> >>     
>
>
>
>
>
>
>
> >>> Hello Brian,
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> thanks for the info. I am a step further, but it cannot load the grammar
>
>
>
>
>
>
>
> >>> files.
>
>
>
>
>
>
>
> >>> I am sending through event_socket:
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> SendMsg
>
>
>
>
>
>
>
> >>> call-command: execute
>
>
>
>
>
>
>
> >>> execute-app-name: detect_speech
>
>
>
>
>
>
>
> >>> execute-app-arg: pocketsphinx yes no
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> However I get the message (also when I am using Pizza demo):
>
>
>
>
>
>
>
> >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
>
>
>
>
>
>
>
> >>> sofia/internal/1000 at sip2.server.com <mailto:1000 at sip2.server.com> Command Execute
>
>
>
>
>
>
>
> >>> detect_speech(pocketsphinx yes no)
>
>
>
>
>
>
>
> >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
>
>
>
>
>
>
>
> >>> pocketsphinx_asr_load_grammar() Can't open language model
>
>
>
>
>
>
>
> >>> /usr/local/freeswitch/grammar/model/communicator.
>
>
>
>
>
>
>
> >>> 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
>
>
>
>
>
>
>
> >>> switch_ivr_detect_speech() Error loading Grammar
>
>
>
>
>
>
>
> >>> 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
>
>
>
>
>
>
>
> >>> pocketsphinx_asr_close() Port Closed.
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> However the grammar files are there:
>
>
>
>
>
>
>
> >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator#
>
>
>
>
>
>
>
> >>> root at sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
>
>
>
>
>
>
>
> >>> total 12752
>
>
>
>
>
>
>
> >>> drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
>
>
>
>
>
>
>
> >>> drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
>
>
>
>
>
>
>
> >>> -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> Any hint?
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> Best regards
>
>
>
>
>
>
>
> >>> Peter
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>> Brian West schrieb:
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>>       
>
>
>
>
>
>
>
> >>>> You can accomplish this .... here is an example using ESL in perl
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>> /b
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>> On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>>
>
>
>
>
>
>
>
> >>>>         
>
>
>
>
>
>
>
> >>>>> Or back to the basics: Is it possible to use pocketsphinx through
>
>
>
>
>
>
>
> >>>>> event
>
>
>
>
>
>
>
> >>>>> socket?
>
>
>
>
>
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>
> >>>>>
>
>
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>
>
> >>>>>
>
>
>
>
>
>
>
> >>>>>           
>
>
>
>
>
>
>
> >>>> _______________________________________________
>
>
>
>
>
>
>
> >>>> 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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>
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> >>>>
>
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> >>>>
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>
>
> >>>>         
>
>
>
>
>
>
>
> >>> _______________________________________________
>
>
>
>
>
>
>
> >>> 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
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>>
>
>
>
>
>
>
>
> >>>       
>
>
>
>
>
>
>
> >> _______________________________________________
>
>
>
>
>
>
>
> >> 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
>
>
>
>
>
>
>
> >>
>
>
>
>
>
>
>
> >>     
>
>
>
>
>
>
>
> >
>
>
>
>
>
>
>
> > _______________________________________________
>
>
>
>
>
>
>
> > 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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> >
>
>
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>
>
>
>
> >   
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
>
>
>
>
>
>
>
> 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
>
>
>
>
>
>
>
>
> ------------------------------------------------------------------------
> *A Good Credit Score is 700 or Above. See yours in just 2 easy steps!
> <http://pr.atwola.com/promoclk/100126575x1219957551x1201325337/aol?redir=http:%2F%2Fwww.freecreditreport.com%2Fpm%2Fdefault.aspx%3Fsc%3D668072%26hmpgID%3D62%26bcd%3DfebemailfooterNO62>*
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 brian at freeswitch.org  Tue Mar  3 06:22:42 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 08:22:42 -0600
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49ACE742.5090809@ewetel.de>
References: <49A7F393.6080406@ewetel.de>
	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>
	<49ACE742.5090809@ewetel.de>
Message-ID: <A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>

That is exactly what it does.  I have confirmed it works can you  
please file a jira with examples and info of what your experiencing?

/b

On Mar 3, 2009, at 2:16 AM, Helmut Kuper wrote:

> Hi Mike,
>
> no, I want just one file in stereo, where left channel is caller and  
> right channel is callee.
>
> regards
> Helmut



From brian at freeswitch.org  Tue Mar  3 06:23:26 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 08:23:26 -0600
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49AD0F7B.7000802@gmx.net>
References: <49AC6EE4.9080509@gmx.net>
	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>
	<49AD0F7B.7000802@gmx.net>
Message-ID: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>

It works if you have the latest SVN with the new grammar files in jsgf  
format.  http://www.bkw.org/pizza_gram.tar.gz


/b

On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote:

> Thanks Mark,
>
> I now switched back to rev. 10003 and the Pizza works again.
>
> Best regards
> Petere



From brian at freeswitch.org  Tue Mar  3 06:24:07 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 08:24:07 -0600
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49AC6EE4.9080509@gmx.net>
References: <49A92BAE.4090907@gmx.net>	<F2465857-6E68-4405-A5C8-B1A57E495AA3@freeswitch.org>	<49AB0BCD.8030108@gmx.net>
	<49AC4233.6060506@gmx.net>
	<92e7d2090903021443v19842db1r7e5342e0bf57e953@mail.gmail.com>
	<49AC6EE4.9080509@gmx.net>
Message-ID: <D9DFDF3E-8890-42BB-AE9B-C2E55F8C4A2E@freeswitch.org>

Looks like the acoustical model wasn't installed... you might need to  
remove all references of pocketsphinx and sphinxbase from libs and let  
it redownload them all.

/b

On Mar 2, 2009, at 5:42 PM, Peter P GMX wrote:

> Thanks Addison.
> The Pizza files are there (as mentionned is it a copy of an already
> working system).
> In fact freeswitch is complaning about
> /usr/local/freeswitch/grammar/model/communicator which he cannot load
>
> So somehow freeswitch is not willing to open the files, but I have no
> clue why. So any hints are welcome.
>
> Best regards
> Peter



From helmut.kuper at ewetel.de  Tue Mar  3 07:24:26 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Tue, 03 Mar 2009 16:24:26 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>
	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>
Message-ID: <49AD4BAA.8090208@ewetel.de>

Hello Brian,

you are right. It works, had to use a different player. But caller is
also there much louder than callee. Any way to tune this?

regards
helmut
 

On 03.03.2009 15:22, Brian West wrote:
> That is exactly what it does.  I have confirmed it works can you  
> please file a jira with examples and info of what your experiencing?



From brian at freeswitch.org  Tue Mar  3 07:40:25 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 09:40:25 -0600
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49AD4BAA.8090208@ewetel.de>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>
	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>
	<49AD4BAA.8090208@ewetel.de>
Message-ID: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>

Thats going to depend on a lot of issues... are you on SVN trunk?   
What codecs? What is the path?

/b

On Mar 3, 2009, at 9:24 AM, Helmut Kuper wrote:

> Hello Brian,
>
> you are right. It works, had to use a different player. But caller is
> also there much louder than callee. Any way to tune this?
>
> regards
> helmut



From kerrada2003 at yahoo.com  Tue Mar  3 07:53:45 2009
From: kerrada2003 at yahoo.com (Ali Al-Rubaie)
Date: Tue, 3 Mar 2009 07:53:45 -0800 (PST)
Subject: [Freeswitch-users] Mandatory IE missing
In-Reply-To: <mailman.6098.1236078493.18013.freeswitch-users@lists.freeswitch.org>
Message-ID: <590483.92469.qm@web33703.mail.mud.yahoo.com>


Hi,

During proxy authentication, I got "mandatory IE missing" error in the response as shown below. How this error can be resolved?





recv 585 bytes from udp/[209.82.10.250]:1898 at
15:31:39.933782:

??
------------------------------------------------------------------------

?? INVITE
sip:9999 at 209.82.10.235 SIP/2.0

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
sip:9999 at 209.82.10.235

?? Contact:
sip:1002 at 209.82.10.250:1091

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988828
INVITE

?? Content-Length: 187

?? Content-Type:
application/sdp

?? User-Agent: HelpCaster
SoftPHONE

?? Supported:
com.hearme.mux

?

?? v=0

?? o=HelpCaster
153553387 153553387 IN IP4 192.168.10.31

?? s=HelpCaster

?? c=IN IP4
209.82.10.250

?? t=0 0

?? m=audio 8002
RTP/AVP 0 4 101

?? a=rtpmap:101
telephone-event/8000

?? a=fmtp:101 0-15

?? ------------------------------------------------------------------------

send 294 bytes to udp/[209.82.10.250]:1091 at
15:31:39.934100:

??
------------------------------------------------------------------------

?? SIP/2.0 100 Trying

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
sip:9999 at 209.82.10.235

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988828
INVITE

?? User-Agent:
FreeSWITCH-mod_sofia/1.0.2-hacked

?? Content-Length: 0

?

??
------------------------------------------------------------------------

send 746 bytes to udp/[209.82.10.250]:1091 at
15:31:39.938913:

??
------------------------------------------------------------------------

?? SIP/2.0 407 Proxy
Authentication Required

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
<sip:9999 at 209.82.10.235>;tag=K83mZmZH8g3Hr

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988828
INVITE

?? User-Agent:
FreeSWITCH-mod_sofia/1.0.2-hacked

?? Accept:
application/sdp

?? Allow: INVITE, ACK,
BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE,
REGISTER, INFO, PUBLISH

?? Supported: timer,
precondition, path, replaces

?? Allow-Events: talk,
presence, dialog, call-info, sla, include-session-description, presence.winfo,
message-summary, refer

?? Proxy-Authenticate:
Digest realm="209.82.10.235",
nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54"

?? Content-Length: 0

?

??
------------------------------------------------------------------------

recv 280 bytes from udp/[209.82.10.250]:1898 at
15:31:39.949517:

??
------------------------------------------------------------------------

?? ACK
sip:9999 at 209.82.10.235 SIP/2.0

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
<sip:9999 at 209.82.10.235>;tag=K83mZmZH8g3Hr

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988828 ACK

?? Content-Length: 0

?

??
------------------------------------------------------------------------

recv 584 bytes from udp/[209.82.10.250]:1898 at
15:31:39.953137:

??
------------------------------------------------------------------------

?? INVITE
sip:9999 at 209.82.10.235 SIP/2.0

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
sip:9999 at 209.82.10.235

?? Contact:
sip:1002 at 209.82.10.250:1091

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988829
INVITE

?? Content-Length: 187

?? Content-Type:
application/sdp

?? User-Agent:
HelpCaster SoftPHONE

?? Supported:
com.hearme.mux

??
Proxy-Authorization:? Digest
username="1002",realm="209.82.10.235",nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response="

d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235"

?

??
------------------------------------------------------------------------

send 294 bytes to udp/[209.82.10.250]:1091 at
15:31:39.953336:

??
------------------------------------------------------------------------

?? SIP/2.0 100 Trying

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
sip:9999 at 209.82.10.235

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988829
INVITE

?? User-Agent:
FreeSWITCH-mod_sofia/1.0.2-hacked

?? Content-Length: 0

?

??
------------------------------------------------------------------------

send 696 bytes to udp/[209.82.10.250]:1091 at
15:31:39.956016:

??
------------------------------------------------------------------------

?? SIP/2.0 480
Temporarily Unavailable

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
<sip:9999 at 209.82.10.235>;tag=mHXD1FgN5SS4K

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988829
INVITE

?? User-Agent:
FreeSWITCH-mod_sofia/1.0.2-hacked

?? Accept:
application/sdp

?? Allow: INVITE, ACK,
BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE,
REGISTER, INFO, PUBLISH

?? Supported: timer,
precondition, path, replaces

?? Allow-Events: talk,
presence, dialog, call-info, sla, include-session-description, presence.winfo,
message-summary, refer

?? Reason:
Q.850;cause=96;text="MANDATORY_IE_MISSING"

?? Content-Length: 0

?

??
------------------------------------------------------------------------

recv 280 bytes from udp/[209.82.10.250]:1898 at
15:31:39.964632:

??
------------------------------------------------------------------------

?? ACK
sip:9999 at 209.82.10.235 SIP/2.0

?? Via: SIP/2.0/UDP
209.82.10.250:1091

?? From: HC-Desktop
<sip:1002 at 209.82.10.235>;tag=55409708

?? To:
<sip:9999 at 209.82.10.235>;tag=mHXD1FgN5SS4K

?? Call-ID:
8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31

?? CSeq: 421988829 ACK

?? Content-Length: 0


Thanks!






      
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From freeswitch at servercorps.com  Tue Mar  3 08:42:01 2009
From: freeswitch at servercorps.com (Addison Martin)
Date: Tue, 3 Mar 2009 10:42:01 -0600
Subject: [Freeswitch-users] First time test set up FreeSwitch and
	SPA3102/SPA3000
In-Reply-To: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com>
References: <9ed22e920903012343u5e50c89fkfa9d1127894199c@mail.gmail.com>
Message-ID: <92e7d2090903030842p509573co969f89a3fb72a8ad@mail.gmail.com>

api,


It may help if we knew a little about what you are trying to do.
Could you explain a bit why you think you need a PBX?  What re your
goals, and objective?  Knowing that may help us get you the
information you need.

Regards,

Nik



On Mon, Mar 2, 2009 at 1:43 AM, Aplayful Idiot
<a.playful.idiot at gmail.com> wrote:
> I have no background in telephony but probably need to use a PBX.
>
> FreeSwitch was recommended by a casual contact so I would like to start
> first by setting up a small test.
>
> I have a SPA3102 attached to the box running FS and to a ordinary phone
> line. I registered SPA in conf/directory/default/line1.xml and it works ok
> but I can't get caller id numbers from incoming calls. All FS sees is
> "line1" which is found in file line1.xml as <variable
> name="effective_caller_id_number" value="line1"/>.
>
> Looking back over the FS wiki, I'm now wondering if the SPA should of been
> set-up as a gateway but reading it is confusing at least to me. Sometimes I
> think the analogue-line-SPA-FS is like a softphone which is registered to an
> extension numbered xml file in conf/directory/default/ but then issues like
> not getting outside incoming caller id's makes me think I've got this all
> wrong.
>
> Can someone help me out with this?
>
> api
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Mar  3 13:17:30 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 3 Mar 2009 15:17:30 -0600
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
In-Reply-To: <1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca>
References: <49AB8562.4050806@laposte.net>
	<E437058E-E930-4060-B71C-80192165614D@jerris.com>
	<49ACD734.7000700@laposte.net>
	<1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca>
Message-ID: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com>

and you may want to update and try trunk to make sure it's not fixed

On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Yes, you may also link (or copy) the .gdbinit file found in the
> support-d folder to your home directory.
> This is going to enable some GDB macros written for FS.
>
> Once thats done you can do the following commands and include them too:
>
> list_sessions
>
> hash_it_str_x session_manager.session_table switch_core_session_t
> channel->state
>
>
> Its important to know that what you see in "show channels" and "show
> calls" is just a DB query to sqlite, Those commands will go directly
> in the core and list those sessions.
>
> Math
>
> On 3-Mar-09, at 2:07 AM, rod wrote:
>
> > Hi Michael,
> >
> > I checked on wiki, is the following the good way to go (sorry I'm not
> > very familiar with your debugging tool).
> >
> > $ gdb bin/freeswitch core.xxx
> >
> > bt
> > bt full
> > thread apply all bt
> > thread apply all bt full
> >
> >
> > If I understand well I have to rerun the tests, as I did not start FS
> > using GDB.
> >
> > regards,
> > rod
> >
> >
> >
> >
> > Michael Jerris wrote:
> >> Could you please post this to jira along with a thread apply all bt
> >> of
> >> a core file taken from the process with the stuck sessions.
> >>
> >> Mike
> >>
> >> On Mar 2, 2009, at 2:06 AM, rod wrote:
> >>
> >>
> >>> Hi All,
> >>>
> >>> I ran some longer tests with FS 1.0.3 acting as an SBC.
> >>> The test machine has the following specs:
> >>>   - Intel Quad Core Q9550
> >>>   - 8GB RAM (far too much from what I saw)
> >>>
> >>> After 3 days running SIPP with 750 simultaneous calls (1500
> >>> channels) at
> >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.
> >>>
> >>> In the CLI, using status command I got this:
> >>>
> >>> freeswitch at internal> status
> >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859
> >>> milliseconds,
> >>> 607 microseconds
> >>> 15817560 session(s) since startup
> >>> 22 session(s) 0/500
> >>>
> >>> But when I use "show channels" or "show calls", I see nothing. So
> >>> I'm
> >>> wondering where are these 22 sessions ?
> >>>
> >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free
> >>> CPUs.
> >>>
> >>> Successful call -->  5271434
> >>> Failed call ---> 1554      (less than 0.03%)
> >>>
> >>> regards,
> >>> rod.
> >>>
> >>>
> >>>
> >>> complete SIPP summary:
> >>>
> >>> ------------------------------ Scenario Screen -------- [1-9]:
> >>> Change
> >>> Screen --
> >>> Call-rate(length)     Port   Total-time  Total-calls  Remote-host
> >>> 50.0(35000 ms)/1.000s   5060  254259.42 s      5273022
> >>> 10.10.10.254:5060(UDP)
> >>>
> >>> 0 new calls during 0.856 s period      7 ms scheduler resolution
> >>> 0 calls (limit 750)                    Peak was 750 calls, after
> >>> 15 s
> >>> 0 Running, 34 Paused, 0 Woken up
> >>> 15544 out-of-call msg (discarded)
> >>> 1 open sockets
> >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP
> >>> rate
> >>> (kB/s)
> >>> 0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate
> >>> (kB/s)
> >>>
> >>>                                Messages  Retrans   Timeout
> >>> Unexpected-Msg
> >>>     INVITE ---------->         5273022   0         0
> >>>        100 <----------         5273022   0                   1554
> >>>        180 <----------         0         0                   0
> >>>        183 <----------         0         0                   0
> >>>        200 <----------  E-RTD1 5271434   0                   0
> >>>        ACK ---------->         5271434   0
> >>>      Pause [    35.0s]         5271434                       0
> >>>        BYE ---------->         5271434   0         0
> >>>        200 <----------         5271434   0                   0
> >>>
> >>> ------------------------------ Test Terminated
> >>> --------------------------------
> >>>
> >>>
> >>> ----------------------------- Statistics Screen ------- [1-9]:
> >>> Change
> >>> Screen --
> >>> Start Time             | 2009-02-27
> >>> 09:11:31
> >>> Last Reset Time        | 2009-03-02
> >>> 07:49:10
> >>> Current Time           | 2009-03-02
> >>> 07:49:11
> >>> -------------------------+---------------------------
> >>> +--------------------------
> >>> Counter Name           | Periodic value            | Cumulative
> >>> value
> >>> -------------------------+---------------------------
> >>> +--------------------------
> >>> Elapsed Time           | 00:00:00:857              |
> >>> 70:37:39:429
> >>> Call Rate              |    0.000 cps              |   20.739
> >>> cps
> >>> -------------------------+---------------------------
> >>> +--------------------------
> >>> Incoming call created  |        0                  |
> >>> 0
> >>> OutGoing call created  |        0                  |
> >>> 5273022
> >>> Total Call created     |                           |
> >>> 5273022
> >>> Current Call           |       34
> >>> |
> >>> -------------------------+---------------------------
> >>> +--------------------------
> >>> Successful call        |        0                  |
> >>> 5271434
> >>> Failed call            |        0                  |
> >>> 1554
> >>> -------------------------+---------------------------
> >>> +--------------------------
> >>> Response Time 1        | 00:00:00:000              |
> >>> 00:00:00:240
> >>> Call Length            | 38:32:13:386              |
> >>> 00:00:36:131
> >>> ------------------------------ Test Terminated
> >>> --------------------------------
> >>>
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Tue Mar  3 14:08:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 3 Mar 2009 17:08:11 -0500
Subject: [Freeswitch-users] Mandatory IE missing
In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com>
References: <590483.92469.qm@web33703.mail.mud.yahoo.com>
Message-ID: <A4227388-3675-4821-B97D-28F3E9F2FE38@jerris.com>

The debug logs should give you more information about what is  
happening here.

Mike

On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie wrote:

> Hi,
>
> During proxy authentication, I got "mandatory IE missing" error in  
> the response as shown below. How this error can be resolved?



From Prometheus001 at gmx.net  Tue Mar  3 17:05:40 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 04 Mar 2009 02:05:40 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>
References: <49AC6EE4.9080509@gmx.net>	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>	<49AD0F7B.7000802@gmx.net>
	<57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>
Message-ID: <49ADD3E4.20408@gmx.net>

Thank you Brian,

I will try this later.

Currently I was happy to get this working on SVN 10003.

As mod_pockesphinx has changed/evolved significantely: Will there also
be major changes in the events I receive through  mod_eventsocket?
I spend some time on parsing the right data out of the eventsocket
interface, and I would just have an idea, if I will have to expect
significant work to do, when I later switch to the current SVN.

Will I need updated grammar files for the other models too?

Best regards
Peter

Brian West schrieb:
> It works if you have the latest SVN with the new grammar files in jsgf  
> format.  http://www.bkw.org/pizza_gram.tar.gz
>
>
> /b
>
> On Mar 3, 2009, at 5:07 AM, Peter P GMX wrote:
>
>   
>> Thanks Mark,
>>
>> I now switched back to rev. 10003 and the Pizza works again.
>>
>> Best regards
>> Petere
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 brian at freeswitch.org  Tue Mar  3 19:00:35 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 21:00:35 -0600
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49ADD3E4.20408@gmx.net>
References: <49AC6EE4.9080509@gmx.net>	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>	<49AD0F7B.7000802@gmx.net>
	<57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>
	<49ADD3E4.20408@gmx.net>
Message-ID: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>

Well you should use ESL then ;)

/b

On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote:

> Thank you Brian,
>
> I will try this later.
>
> Currently I was happy to get this working on SVN 10003.
>
> As mod_pockesphinx has changed/evolved significantely: Will there also
> be major changes in the events I receive through  mod_eventsocket?
> I spend some time on parsing the right data out of the eventsocket
> interface, and I would just have an idea, if I will have to expect
> significant work to do, when I later switch to the current SVN.
>
> Will I need updated grammar files for the other models too?
>
> Best regards
> Peter



From jgarland at jasongarland.com  Tue Mar  3 20:12:45 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Tue, 3 Mar 2009 23:12:45 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] Qt portaudio interface
In-Reply-To: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca>
References: <3A65F8B6-96B4-45EE-80E9-A30AFEED0BB3@avgs.ca>
Message-ID: <BEF8F1BF-B0CA-45D4-B2CF-212162C63E3D@jasongarland.com>

I still think a web based interface would work well and be more cross  
platform. The web app could be served from within FS. http://127.0.0.1:8080/myfancysoftphone/

Sent from my iPhone

On Mar 2, 2009, at 12:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi all,
>
> Anyone interested in contributing to a Qt interface in order to make a
> decent softphone using FS please reply to this thread.
> (also give your availability so we can have a conference call to
> decide stuff)
>
> Thanks,
> Math
>
> _______________________________________________
> Freeswitch-dev mailing list
> Freeswitch-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org


From jgarland at jasongarland.com  Tue Mar  3 20:18:40 2009
From: jgarland at jasongarland.com (Jason Garland)
Date: Tue, 3 Mar 2009 23:18:40 -0500
Subject: [Freeswitch-users] Mandatory IE missing
In-Reply-To: <590483.92469.qm@web33703.mail.mud.yahoo.com>
References: <590483.92469.qm@web33703.mail.mud.yahoo.com>
Message-ID: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com>

Your HelpCaster softphone didn't send any SDP on the second INVITE.  
Fix the softphone or try turning on late codec negotiation in you  
Sofia profile.

Sent from my iPhone

On Mar 3, 2009, at 10:53 AM, Ali Al-Rubaie <kerrada2003 at yahoo.com>  
wrote:

>
> Hi,
>
> During proxy authentication, I got "mandatory IE missing" error in  
> the response as shown below. How this error can be resolved?
>
> recv 585 bytes from udp/[209.82.10.250]:1898 at 15:31:39.933782:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    INVITE sip:9999 at 209.82.10.235 SIP/2.0
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: sip:9999 at 209.82.10.235
>
>    Contact: sip:1002 at 209.82.10.250:1091
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988828 INVITE
>
>    Content-Length: 187
>
>    Content-Type: application/sdp
>
>    User-Agent: HelpCaster SoftPHONE
>
>    Supported: com.hearme.mux
>
>
>
>    v=0
>
>    o=HelpCaster 153553387 153553387 IN IP4 192.168.10.31
>
>    s=HelpCaster
>
>    c=IN IP4 209.82.10.250
>
>    t=0 0
>
>    m=audio 8002 RTP/AVP 0 4 101
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-15
>
>     
> --- 
> ---------------------------------------------------------------------
>
> send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.934100:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    SIP/2.0 100 Trying
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: sip:9999 at 209.82.10.235
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988828 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
>
>    Content-Length: 0
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> send 746 bytes to udp/[209.82.10.250]:1091 at 15:31:39.938913:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    SIP/2.0 407 Proxy Authentication Required
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: <sip:9999 at 209.82.10.235>;tag=K83mZmZH8g3Hr
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988828 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
>
>    Accept: application/sdp
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>
>    Supported: timer, precondition, path, replaces
>
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>
>    Proxy-Authenticate: Digest realm="209.82.10.235",  
> nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54"
>
>    Content-Length: 0
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.949517:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    ACK sip:9999 at 209.82.10.235 SIP/2.0
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: <sip:9999 at 209.82.10.235>;tag=K83mZmZH8g3Hr
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988828 ACK
>
>    Content-Length: 0
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> recv 584 bytes from udp/[209.82.10.250]:1898 at 15:31:39.953137:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    INVITE sip:9999 at 209.82.10.235 SIP/2.0
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: sip:9999 at 209.82.10.235
>
>    Contact: sip:1002 at 209.82.10.250:1091
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988829 INVITE
>
>    Content-Length: 187
>
>    Content-Type: application/sdp
>
>    User-Agent: HelpCaster SoftPHONE
>
>    Supported: com.hearme.mux
>
>    Proxy-Authorization:  Digest  
> username= 
> "1002" 
> ,realm= 
> "209.82.10.235" 
> ,nonce="248a2800-7fd2-4a6f-8082-5d1731f3cf54",response="
>
> d1614cd024acab0b794751285f5cb1fe",uri="sip:9999 at 209.82.10.235"
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> send 294 bytes to udp/[209.82.10.250]:1091 at 15:31:39.953336:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    SIP/2.0 100 Trying
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: sip:9999 at 209.82.10.235
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988829 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
>
>    Content-Length: 0
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> send 696 bytes to udp/[209.82.10.250]:1091 at 15:31:39.956016:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    SIP/2.0 480 Temporarily Unavailable
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: <sip:9999 at 209.82.10.235>;tag=mHXD1FgN5SS4K
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988829 INVITE
>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked
>
>    Accept: application/sdp
>
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>
>    Supported: timer, precondition, path, replaces
>
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>
>    Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING"
>
>    Content-Length: 0
>
>
>
>     
> --- 
> ---------------------------------------------------------------------
>
> recv 280 bytes from udp/[209.82.10.250]:1898 at 15:31:39.964632:
>
>     
> --- 
> ---------------------------------------------------------------------
>
>    ACK sip:9999 at 209.82.10.235 SIP/2.0
>
>    Via: SIP/2.0/UDP 209.82.10.250:1091
>
>    From: HC-Desktop <sip:1002 at 209.82.10.235>;tag=55409708
>
>    To: <sip:9999 at 209.82.10.235>;tag=mHXD1FgN5SS4K
>
>    Call-ID: 8e6df5d2-0367-43d9-bda8-f51a94033473 at 192.168.10.31
>
>    CSeq: 421988829 ACK
>
>    Content-Length: 0
>
>
> Thanks!
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Tue Mar  3 20:21:08 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 3 Mar 2009 22:21:08 -0600
Subject: [Freeswitch-users] Mandatory IE missing
In-Reply-To: <0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com>
References: <590483.92469.qm@web33703.mail.mud.yahoo.com>
	<0744A0FC-7FCE-4662-944D-9FE133142AE1@jasongarland.com>
Message-ID: <3E391AC2-BF15-4987-B143-5CF05855693E@freeswitch.org>

Jason thanks.. you know I think we all missed that one.

/b

On Mar 3, 2009, at 10:18 PM, Jason Garland wrote:

> Your HelpCaster softphone didn't send any SDP on the second INVITE.  
> Fix the softphone or try turning on late codec negotiation in you  
> Sofia profile.
>
> Sent from my iPhone



From kawarod at laposte.net  Tue Mar  3 22:47:21 2009
From: kawarod at laposte.net (rod)
Date: Wed, 04 Mar 2009 10:47:21 +0400
Subject: [Freeswitch-users] Ghost Sessions in CLI after a longterm test
In-Reply-To: <191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com>
References: <49AB8562.4050806@laposte.net>	<E437058E-E930-4060-B71C-80192165614D@jerris.com>	<49ACD734.7000700@laposte.net>	<1C6ECC88-DD99-4C09-8806-74264FDDC234@avgs.ca>
	<191c3a030903031317y55ce2c9bsb4a93518db261973@mail.gmail.com>
Message-ID: <49AE23F9.7050108@laposte.net>

Hi,

I'm already trying this.
I will update this thread next week.

regards.

Anthony Minessale wrote:
> and you may want to update and try trunk to make sure it's not fixed
>
> On Tue, Mar 3, 2009 at 1:56 AM, Mathieu Rene <mrene_lists at avgs.ca 
> <mailto:mrene_lists at avgs.ca>> wrote:
>
>     Yes, you may also link (or copy) the .gdbinit file found in the
>     support-d folder to your home directory.
>     This is going to enable some GDB macros written for FS.
>
>     Once thats done you can do the following commands and include them
>     too:
>
>     list_sessions
>
>     hash_it_str_x session_manager.session_table switch_core_session_t
>     channel->state
>
>
>     Its important to know that what you see in "show channels" and "show
>     calls" is just a DB query to sqlite, Those commands will go directly
>     in the core and list those sessions.
>
>     Math
>
>     On 3-Mar-09, at 2:07 AM, rod wrote:
>
>     > Hi Michael,
>     >
>     > I checked on wiki, is the following the good way to go (sorry
>     I'm not
>     > very familiar with your debugging tool).
>     >
>     > $ gdb bin/freeswitch core.xxx
>     >
>     > bt
>     > bt full
>     > thread apply all bt
>     > thread apply all bt full
>     >
>     >
>     > If I understand well I have to rerun the tests, as I did not
>     start FS
>     > using GDB.
>     >
>     > regards,
>     > rod
>     >
>     >
>     >
>     >
>     > Michael Jerris wrote:
>     >> Could you please post this to jira along with a thread apply all bt
>     >> of
>     >> a core file taken from the process with the stuck sessions.
>     >>
>     >> Mike
>     >>
>     >> On Mar 2, 2009, at 2:06 AM, rod wrote:
>     >>
>     >>
>     >>> Hi All,
>     >>>
>     >>> I ran some longer tests with FS 1.0.3 acting as an SBC.
>     >>> The test machine has the following specs:
>     >>>   - Intel Quad Core Q9550
>     >>>   - 8GB RAM (far too much from what I saw)
>     >>>
>     >>> After 3 days running SIPP with 750 simultaneous calls (1500
>     >>> channels) at
>     >>> 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.
>     >>>
>     >>> In the CLI, using status command I got this:
>     >>>
>     >>> freeswitch at internal> status
>     >>> UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859
>     >>> milliseconds,
>     >>> 607 microseconds
>     >>> 15817560 session(s) since startup
>     >>> 22 session(s) 0/500
>     >>>
>     >>> But when I use "show channels" or "show calls", I see nothing. So
>     >>> I'm
>     >>> wondering where are these 22 sessions ?
>     >>>
>     >>> FYI, FS has run flawlessly with 750 sim. calls with 25-30% free
>     >>> CPUs.
>     >>>
>     >>> Successful call -->  5271434
>     >>> Failed call ---> 1554      (less than 0.03%)
>     >>>
>     >>> regards,
>     >>> rod.
>     >>>
>     >>>
>     >>>
>     >>> complete SIPP summary:
>     >>>
>     >>> ------------------------------ Scenario Screen -------- [1-9]:
>     >>> Change
>     >>> Screen --
>     >>> Call-rate(length)     Port   Total-time  Total-calls  Remote-host
>     >>> 50.0(35000 ms)/1.000s   5060  254259.42 s      5273022
>     >>> 10.10.10.254:5060(UDP)
>     >>>
>     >>> 0 new calls during 0.856 s period      7 ms scheduler resolution
>     >>> 0 calls (limit 750)                    Peak was 750 calls, after
>     >>> 15 s
>     >>> 0 Running, 34 Paused, 0 Woken up
>     >>> 15544 out-of-call msg (discarded)
>     >>> 1 open sockets
>     >>> 9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP
>     >>> rate
>     >>> (kB/s)
>     >>> 0 Total echo RTP pckts 2nd stream      0.000 last period RTP rate
>     >>> (kB/s)
>     >>>
>     >>>                                Messages  Retrans   Timeout
>     >>> Unexpected-Msg
>     >>>     INVITE ---------->         5273022   0         0
>     >>>        100 <----------         5273022   0                   1554
>     >>>        180 <----------         0         0                   0
>     >>>        183 <----------         0         0                   0
>     >>>        200 <----------  E-RTD1 5271434   0                   0
>     >>>        ACK ---------->         5271434   0
>     >>>      Pause [    35.0s]         5271434                       0
>     >>>        BYE ---------->         5271434   0         0
>     >>>        200 <----------         5271434   0                   0
>     >>>
>     >>> ------------------------------ Test Terminated
>     >>> --------------------------------
>     >>>
>     >>>
>     >>> ----------------------------- Statistics Screen ------- [1-9]:
>     >>> Change
>     >>> Screen --
>     >>> Start Time             | 2009-02-27
>     >>> 09:11:31
>     >>> Last Reset Time        | 2009-03-02
>     >>> 07:49:10
>     >>> Current Time           | 2009-03-02
>     >>> 07:49:11
>     >>> -------------------------+---------------------------
>     >>> +--------------------------
>     >>> Counter Name           | Periodic value            | Cumulative
>     >>> value
>     >>> -------------------------+---------------------------
>     >>> +--------------------------
>     >>> Elapsed Time           | 00:00:00:857              |
>     >>> 70:37:39:429
>     >>> Call Rate              |    0.000 cps              |   20.739
>     >>> cps
>     >>> -------------------------+---------------------------
>     >>> +--------------------------
>     >>> Incoming call created  |        0                  |
>     >>> 0
>     >>> OutGoing call created  |        0                  |
>     >>> 5273022
>     >>> Total Call created     |                           |
>     >>> 5273022
>     >>> Current Call           |       34
>     >>> |
>     >>> -------------------------+---------------------------
>     >>> +--------------------------
>     >>> Successful call        |        0                  |
>     >>> 5271434
>     >>> Failed call            |        0                  |
>     >>> 1554
>     >>> -------------------------+---------------------------
>     >>> +--------------------------
>     >>> Response Time 1        | 00:00:00:000              |
>     >>> 00:00:00:240
>     >>> Call Length            | 38:32:13:386              |
>     >>> 00:00:36:131
>     >>> ------------------------------ Test Terminated
>     >>> --------------------------------
>     >>>
>     >>
>     >>
>     >> _______________________________________________
>     >> 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
>     >>
>     >>
>     >>
>     >
>     > _______________________________________________
>     > 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
>
>
>     _______________________________________________
>     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/
>
> 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>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mszlazak at aol.com  Tue Mar  3 23:24:10 2009
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Wed, 04 Mar 2009 02:24:10 -0500
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>
References: <49AC6EE4.9080509@gmx.net>	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>	<49AD0F7B.7000802@gmx.net><57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org><49ADD3E4.20408@gmx.net>
	<8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>
Message-ID: <8CB6AB158D38B9A-A74-24FC@WEBMAIL-MA07.sysops.aol.com>


 Brian,

Peter says: 

"mod_pockesphinx has changed/evolved significantely"

Since this seems to be coming without any warning, what specifically are all these and future changes and why are they happening?  



 Mark.


 

-----Original Message-----
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, 3 Mar 2009 7:00 pm
Subject: Re: [Freeswitch-users] pocketsphinx and event socket










Well you should use ESL then ;)

/b

On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote:

> Thank you Brian,
>
> I will try this later.
>
> Currently I was happy to get this working on SVN 10003.
>
> As : Will there also
> be major changes in the events I receive through  mod_eventsocket?
> I spend some time on parsing the right data out of the eventsocket
> interface, and I would just have an idea, if I will have to expect
> significant work to do, when I later switch to the current SVN.
>
> Will I need updated grammar files for the other models too?
>
> Best regards
> Peter


_______________________________________________
Freeswitch-users mailing 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 Claudio.Cavalera at italtel.it  Wed Mar  4 03:41:53 2009
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 4 Mar 2009 12:41:53 +0100
Subject: [Freeswitch-users] howto master logging in fs_cli
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969795@BESONE.corp.dom>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC09697DA@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> freeswitch-users-bounces at lists.freeswitch.org wrote:
>> Cavalera Claudio Luigi <Claudio.Cavalera at italtel.it> wrote:
>>> So could you please someone more expert with me clarify the
>>> difference between -d and -l options so that I can update the wiki
>>> which is now wrong/incomplete ?
>>> 
>>>   -l, --loglevel=command          Log Level
>>>   -q, --quiet                             Disable logging
>>>   -d, --debug=level               Debug Level (0 - 7)
>> 
>> The difference is that -d controls the level of debugging output
>> generated by fs_cli itself. The log level controls which log messages
>> from your running FreeSWITCH daemon are printed to the fs_cli
>> console. 


Could you please explain why if I issue this command in fs_cli:
"fsctl loglevel warning"
to lower the log level in output file freeswitch.log
but then I can't have level debug in fs_cli even if I re-set it with
"/log debug"

The fsctl loglevel (which I guess is the same as the global loglevel in
switch.conf.xml) must always be higher than loglevel in fs_cli?
If correct, is this true also for the console_loglevel ?

Thanks,
Claudio


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From helmut.kuper at ewetel.de  Wed Mar  4 04:28:52 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 04 Mar 2009 13:28:52 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>
	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>
Message-ID: <49AE7404.8000905@ewetel.de>

Hi,

well, both extensions are direct connected to FS, so we have an internal
call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP
crypt AES 32, both phones are within the same LAN. FS is in a separate
LAN. Energy levels for speakers and micros on both snom phones are the same.

I use latest SVN trunk.

regards
Helmut


On 03.03.2009 16:40, Brian West wrote:
> Thats going to depend on a lot of issues... are you on SVN trunk?   
> What codecs? What is the path?
>
> /b
>



From mrene_lists at avgs.ca  Wed Mar  4 04:49:06 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 4 Mar 2009 07:49:06 -0500
Subject: [Freeswitch-users] howto master logging in fs_cli
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC09697DA@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC09697DA@BESONE.corp.dom>
Message-ID: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca>

fsctl loglevel [xxx] tells the core to ignore log messages not at  
least that level. They wont get logged, you wont see them on the  
console and no events will be generated.

console loglevel [xxx] (on the "real" console) tells mod_console to do  
that filtering.
/log [xxx] tells fs_cli to do the filtering.

It all depends where you filter it, if you want to change the  
logfile's level without affecting anything else, I recommand you edit  
logfile.conf.xml

Math

On 4-Mar-09, at 6:41 AM, Cavalera Claudio Luigi wrote:

> freeswitch-users-bounces at lists.freeswitch.org wrote:
>> freeswitch-users-bounces at lists.freeswitch.org wrote:
>>> Cavalera Claudio Luigi <Claudio.Cavalera at italtel.it> wrote:
>>>> So could you please someone more expert with me clarify the
>>>> difference between -d and -l options so that I can update the wiki
>>>> which is now wrong/incomplete ?
>>>>
>>>>  -l, --loglevel=command          Log Level
>>>>  -q, --quiet                             Disable logging
>>>>  -d, --debug=level               Debug Level (0 - 7)
>>>
>>> The difference is that -d controls the level of debugging output
>>> generated by fs_cli itself. The log level controls which log  
>>> messages
>>> from your running FreeSWITCH daemon are printed to the fs_cli
>>> console.
>
>
> Could you please explain why if I issue this command in fs_cli:
> "fsctl loglevel warning"
> to lower the log level in output file freeswitch.log
> but then I can't have level debug in fs_cli even if I re-set it with
> "/log debug"
>
> The fsctl loglevel (which I guess is the same as the global loglevel  
> in
> switch.conf.xml) must always be higher than loglevel in fs_cli?
> If correct, is this true also for the console_loglevel ?
>
> Thanks,
> Claudio
>
>
> Internet Email Confidentiality Footer
> -----------------------------------------------------------------------------------------------------
> La presente comunicazione, con le informazioni in essa contenute e  
> ogni documento o file allegato, e' rivolta unicamente alla/e persona/ 
> e cui e' indirizzata ed alle altre da questa autorizzata/e a  
> riceverla. Se non siete i destinatari/autorizzati siete avvisati che  
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> sul contenuto di tali informazioni e' vietata e potrebbe essere  
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From helmut.kuper at ewetel.de  Wed Mar  4 05:08:19 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 04 Mar 2009 14:08:19 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49AE7404.8000905@ewetel.de>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>
	<49AE7404.8000905@ewetel.de>
Message-ID: <49AE7D43.9070409@ewetel.de>

Hello,

I just did a further test to nail down the volume problem with recording
a call. Caller as well as callee recorded the session in stereo. I got
two files in "recordings". In both files, the party which starts the
recording seems to be louder and more clear as the opposite party. It
seems to me, that the party which didn't start the recording was
transcoded to a different codec (guess g711). Remember: Both partys are
using g722 which was confirmed by the phone's display.


regards
helmut
 

On 04.03.2009 13:28, Helmut Kuper wrote:
> Hi,
>
> well, both extensions are direct connected to FS, so we have an internal
> call. Caller is snom 370, callee is snom 820, codecs are G.722 with RTP
> crypt AES 32, both phones are within the same LAN. FS is in a separate
> LAN. Energy levels for speakers and micros on both snom phones are the same.
>
> I use latest SVN trunk.
>
> regards
> Helmut



From Claudio.Cavalera at italtel.it  Wed Mar  4 07:03:46 2009
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Wed, 4 Mar 2009 16:03:46 +0100
Subject: [Freeswitch-users] howto master logging in fs_cli
In-Reply-To: <0321F1A5-F307-465E-8204-FE9FF4842900@avgs.ca>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC09697DD@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> fsctl loglevel [xxx] tells the core to ignore log messages not at
> least that level. They wont get logged, you wont see them on the
> console and no events will be generated.
> 
> console loglevel [xxx] (on the "real" console) tells
> mod_console to do
> that filtering.
> /log [xxx] tells fs_cli to do the filtering.
> 
> It all depends where you filter it, if you want to change the
> logfile's level without affecting anything else, I recommand
> you edit
> logfile.conf.xml
> 
> Math


Ah ok, so it's better to keep switch.conf.xml to debug and then to
adjust the single sources to lower logging levels.
I have also put these lines in a rc.freeswitch script before starting
freeswitch
export SOFIA_DEBUG=9
export NUA_DEBUG=9
export NTA_DEBUG=9
export TPORT_DEBUG=9
export TPORT_LOG=1
becase I wanted to have all the sofia output redirected to a different
logfile but it does not work.

Instead if on any console I use fs_cli I see there the sofia stuff, how
does that work?

Thanks,
Claudio
Internet E. Mail Confidentiality Footer
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La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. 

This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. 


From brian at freeswitch.org  Wed Mar  4 07:13:01 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 Mar 2009 09:13:01 -0600
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49AE7D43.9070409@ewetel.de>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>
	<49AE7404.8000905@ewetel.de> <49AE7D43.9070409@ewetel.de>
Message-ID: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org>

Can you email me a sample recording please?

/b

On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote:

> Hello,
>
> I just did a further test to nail down the volume problem with  
> recording
> a call. Caller as well as callee recorded the session in stereo. I got
> two files in "recordings". In both files, the party which starts the
> recording seems to be louder and more clear as the opposite party. It
> seems to me, that the party which didn't start the recording was
> transcoded to a different codec (guess g711). Remember: Both partys  
> are
> using g722 which was confirmed by the phone's display.
>
>
> regards
> helmut



From helmut.kuper at ewetel.de  Wed Mar  4 07:37:38 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 04 Mar 2009 16:37:38 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>	<49AE7404.8000905@ewetel.de>
	<49AE7D43.9070409@ewetel.de>
	<3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org>
Message-ID: <49AEA042.3090908@ewetel.de>

Hi Brian,

both files are on their way ... Quite big (18MB)

regards
Helmut

On 04.03.2009 16:13, Brian West wrote:
> Can you email me a sample recording please?
>
> /b
>
> On Mar 4, 2009, at 7:08 AM, Helmut Kuper wrote:
>
>   
>> Hello,
>>
>> I just did a further test to nail down the volume problem with  
>> recording
>> a call. Caller as well as callee recorded the session in stereo. I got
>> two files in "recordings". In both files, the party which starts the
>> recording seems to be louder and more clear as the opposite party. It
>> seems to me, that the party which didn't start the recording was
>> transcoded to a different codec (guess g711). Remember: Both partys  
>> are
>> using g722 which was confirmed by the phone's display.
>>
>>
>> regards
>> helmut
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Wed Mar  4 07:43:51 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 Mar 2009 09:43:51 -0600
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <49AEA042.3090908@ewetel.de>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>	<49AE7404.8000905@ewetel.de>
	<49AE7D43.9070409@ewetel.de>
	<3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org>
	<49AEA042.3090908@ewetel.de>
Message-ID: <D719BF4A-9D11-4C29-B06A-922DC89C6FC7@freeswitch.org>

Already received them.. Opened them in Audacity.. compared them...  
listened to them a couple of times... and they sound fine to me.... So  
answer the two questions I asked in the private email and we'll see ;)

/b

On Mar 4, 2009, at 9:37 AM, Helmut Kuper wrote:

> Hi Brian,
>
> both files are on their way ... Quite big (18MB)
>
> regards
> Helmut



From helmut.kuper at ewetel.de  Wed Mar  4 08:54:14 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 04 Mar 2009 17:54:14 +0100
Subject: [Freeswitch-users] How to send CUSTOM event via event socket?
Message-ID: <49AEB236.10201@ewetel.de>

Hello,

how to send a CUSTOM event via event socket? Currently I send this:

sendevent CUSTOM\n
Event-Name: CUSTOM\n
Event-Subclass: myevent::snom-aurl\n
Snom-Event: OFFHOOK\n
\n


I got an +OK reply from FS. I subscribed to ALL event and get only this:

Content-Length: 516
Content-Type: text/event-plain

Event-Name: COMMAND
Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d
FreeSWITCH-Hostname: ippbx-prod-node0
FreeSWITCH-IPv4: 85.16.246.5
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-03-04%2017%3A42%3A35
Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT
Event-Date-Timestamp: 1236184955199959
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: read_packet
Event-Calling-Line-Number: 1045
Command: sendevent%20CUSTOM
Event-Name: CUSTOM
Event-Subclass: ewetel%3A%3Asnom-aurl
Snom-Event: OFFHOOK

But there is no CUSTOM event received.

regards
helmut




From fax at virgintechnologies.com  Wed Mar  4 10:45:45 2009
From: fax at virgintechnologies.com (Justin Miller)
Date: Wed, 04 Mar 2009 18:45:45 +0000
Subject: [Freeswitch-users] separate gateways SIP and RTP stream
Message-ID: <W894969492327101236192345@webmail13>

I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060.
Thank you
Justin

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From mrene_lists at avgs.ca  Wed Mar  4 10:47:47 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 4 Mar 2009 13:47:47 -0500
Subject: [Freeswitch-users] separate gateways SIP and RTP stream
In-Reply-To: <W894969492327101236192345@webmail13>
References: <W894969492327101236192345@webmail13>
Message-ID: <1AE3D5A9-6753-4005-947A-A92B16F55DF9@avgs.ca>

Using different IPs for signalling and media is perfectly legal, thats  
why the SDP contains information on where the media should be sent..


Math

On 4-Mar-09, at 1:45 PM, Justin Miller wrote:

> I'm setting up a test SIP trunk with Allstream.  They are using  
> separate gateway IPs for SIP signalling and RTP streams.  Has anyone  
> done this with Freeswitch?  I was planning on setting them up as an  
> extension in the internal profile since they use SIP port 5060.
> Thank you
> Justin
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 fax at virgintechnologies.com  Wed Mar  4 11:25:03 2009
From: fax at virgintechnologies.com (Justin Miller)
Date: Wed, 04 Mar 2009 19:25:03 +0000
Subject: [Freeswitch-users] separate gateways SIP and RTP stream
Message-ID: <W943027299180121236194703@webmail27>

So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream.
-----Original Message-----
From: Mathieu Rene [mailto:mrene_lists at avgs.ca]
Sent: Wednesday, March 4, 2009 11:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream

Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. 



Math

On 4-Mar-09, at 1:45 PM, Justin Miller wrote:

I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060.
Thank you
Justin

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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 mrene_lists at avgs.ca  Wed Mar  4 11:26:10 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 4 Mar 2009 14:26:10 -0500
Subject: [Freeswitch-users] separate gateways SIP and RTP stream
In-Reply-To: <W943027299180121236194703@webmail27>
References: <W943027299180121236194703@webmail27>
Message-ID: <259504C8-C9CA-4A0E-98E6-DC091734D356@avgs.ca>

You don't. The SIP protocol already takes care of that, just make sure  
you open your firewall properly.

On 4-Mar-09, at 2:25 PM, Justin Miller wrote:

> So is there an "RTP_proxy" parameter that can be set for the  
> gateway, or in my case, the user extension?  How would I define the  
> separate IP for the RTP stream.
> -----Original Message-----
> From: Mathieu Rene [mailto:mrene_lists at avgs.ca]
> Sent: Wednesday, March 4, 2009 11:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> Using different IPs for signalling and media is perfectly legal,  
> thats why the SDP contains information on where the media should be  
> sent..
>
>
> Math
>
> On 4-Mar-09, at 1:45 PM, Justin Miller wrote:
>
>> I'm setting up a test SIP trunk with Allstream.  They are using  
>> separate gateway IPs for SIP signalling and RTP streams.  Has  
>> anyone done this with Freeswitch?  I was planning on setting them  
>> up as an extension in the internal profile since they use SIP port  
>> 5060.
>> Thank you
>> Justin
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at 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 mrene_lists at avgs.ca  Wed Mar  4 11:27:43 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 4 Mar 2009 14:27:43 -0500
Subject: [Freeswitch-users] How to send CUSTOM event via event socket?
In-Reply-To: <49AEB236.10201@ewetel.de>
References: <49AEB236.10201@ewetel.de>
Message-ID: <0C5EA2F5-9B6D-4D00-98D1-2DD60C8E8F59@avgs.ca>


On 4-Mar-09, at 11:54 AM, Helmut Kuper wrote:


> Event-Name: CUSTOM
> Event-Subclass: ewetel%3A%3Asnom-aurl
> Snom-Event: OFFHOOK

^^
mod_event_socket loads up commands it receives in an event struct,  
when you do a "sendevent", it changes the eventid to be changed to the  
one you specified, unfortunately it doesnt remove the first Event-Name  
header.

Open a JIRA if thats what you want.

Math

> Hello,
>
> how to send a CUSTOM event via event socket? Currently I send this:
>
> sendevent CUSTOM\n
> Event-Name: CUSTOM\n
> Event-Subclass: myevent::snom-aurl\n
> Snom-Event: OFFHOOK\n
> \n
>
>
> I got an +OK reply from FS. I subscribed to ALL event and get only  
> this:
>
> Content-Length: 516
> Content-Type: text/event-plain
>
> Event-Name: COMMAND
> Core-UUID: 7b2b75a4-08ca-11de-b1f3-3d0cddd2708d
> FreeSWITCH-Hostname: ippbx-prod-node0
> FreeSWITCH-IPv4: 85.16.246.5
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-03-04%2017%3A42%3A35
> Event-Date-GMT: Wed,%2004%20Mar%202009%2016%3A42%3A35%20GMT
> Event-Date-Timestamp: 1236184955199959
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: read_packet
> Event-Calling-Line-Number: 1045
> Command: sendevent%20CUSTOM
> Event-Name: CUSTOM
> Event-Subclass: ewetel%3A%3Asnom-aurl
> Snom-Event: OFFHOOK
>
> But there is no CUSTOM event received.
>
> regards
> helmut
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 fax at virgintechnologies.com  Wed Mar  4 12:25:16 2009
From: fax at virgintechnologies.com (Justin Miller)
Date: Wed, 04 Mar 2009 20:25:16 +0000
Subject: [Freeswitch-users] separate gateways SIP and RTP stream
Message-ID: <W968972186095461236198316@webmail30>

Ok, got it. Thank you.
-----Original Message-----
From: Mathieu Rene [mailto:mrene_lists at avgs.ca]
Sent: Wednesday, March 4, 2009 12:26 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream

You don't. The SIP protocol already takes care of that, just make sure you open your firewall properly. 
On 4-Mar-09, at 2:25 PM, Justin Miller wrote:

So is there an "RTP_proxy" parameter that can be set for the gateway, or in my case, the user extension? How would I define the separate IP for the RTP stream.
-----Original Message-----
From: Mathieu Rene [mailto:mrene_lists at avgs.ca]
Sent: Wednesday, March 4, 2009 11:47 AM
To:freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] separate gateways SIP and RTP stream

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Using different IPs for signalling and media is perfectly legal, thats why the SDP contains information on where the media should be sent.. 



Math

On 4-Mar-09, at 1:45 PM, Justin Miller wrote:

I'm setting up a test SIP trunk with Allstream. They are using separate gateway IPs for SIP signalling and RTP streams. Has anyone done this with Freeswitch? I was planning on setting them up as an extension in the internal profile since they use SIP port 5060.
Thank you
Justin

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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 dule.maillist at gmail.com  Wed Mar  4 15:58:19 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Wed, 4 Mar 2009 18:58:19 -0500
Subject: [Freeswitch-users] SIP registration/retry/authorization problem
In-Reply-To: <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org>
References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com>
	<286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org>
	<24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com>
	<980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org>
Message-ID: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com>

What does <param name="auto-restart" value="false"/> do exactly? When do you
need it?
I'll wikify the response.

Thanks,
Dan

On Tue, Feb 10, 2009 at 8:48 PM, Brian West <brian at freeswitch.org> wrote:

> I highly recommend you wipe the box/install and install from Scratch
> using SVN trunk
>
>
> /b
> - Show quoted text -
>
> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
>
> > I'm not able to find any documentation on this setting. I think it may
> > be newer than my version of FreeSwitch (1.0). What does it do?
> >
> > Thanks,
> > - Jesse
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mrene_lists at avgs.ca  Wed Mar  4 16:00:33 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 4 Mar 2009 19:00:33 -0500
Subject: [Freeswitch-users] SIP registration/retry/authorization problem
In-Reply-To: <914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com>
References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com>
	<286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org>
	<24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com>
	<980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org>
	<914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com>
Message-ID: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca>

it auto restarts the profile when a network address change is detected.

On 4-Mar-09, at 6:58 PM, Dan Le wrote:

> What does <param name="auto-restart" value="false"/> do exactly?  
> When do you need it?
>
> I'll wikify the response.
>
> Thanks,
> Dan
>
> On Tue, Feb 10, 2009 at 8:48 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I highly recommend you wipe the box/install and install from Scratch
> using SVN trunk
>
>
> /b
> - Show quoted text -
>
> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
>
> > I'm not able to find any documentation on this setting. I think it  
> may
> > be newer than my version of FreeSwitch (1.0). What does it do?
> >
> > Thanks,
> > - Jesse
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 dule.maillist at gmail.com  Wed Mar  4 18:54:50 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Wed, 4 Mar 2009 21:54:50 -0500
Subject: [Freeswitch-users] SIP registration/retry/authorization problem
In-Reply-To: <226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca>
References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com>
	<286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org>
	<24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com>
	<980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org>
	<914fc92a0903041558x1588cfacx58509eb5af160875@mail.gmail.com>
	<226909BF-C3DC-4172-803A-34329C31CC63@avgs.ca>
Message-ID: <914fc92a0903041854k4f5a95e2j3692ca09c0582309@mail.gmail.com>

Thanks, wiki'd as:
http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
Dan

On Wed, Mar 4, 2009 at 7:00 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> it auto restarts the profile when a network address change is detected.-
> Show quoted text -
>
> On 4-Mar-09, at 6:58 PM, Dan Le wrote:
>
> What does <param name="auto-restart" value="false"/> do exactly? When do
> you need it?
> I'll wikify the response.
>
> Thanks,
> Dan
>
> On Tue, Feb 10, 2009 at 8:48 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I highly recommend you wipe the box/install and install from Scratch
>> using SVN trunk
>>
>>
>> /b
>> - Show quoted text -
>>
>> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
>>
>> > I'm not able to find any documentation on this setting. I think it may
>> > be newer than my version of FreeSwitch (1.0). What does it do?
>> >
>> > Thanks,
>> > - Jesse
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Mar  4 19:31:24 2009
From: gerry at pstn2.net (Gerry Hull)
Date: Wed, 4 Mar 2009 22:31:24 -0500
Subject: [Freeswitch-users] mod_unistim?
Message-ID: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com>

I hear rumors that someone is porting chan_unistim to mod_unistim for
FreeSwitch??  I hope so -- I use this on my Asterisk box and would love to
use it with FS.   There are TONS of i2004 phones on the surplus market these
days... I've been buying NOS i2004s, virgin, for less than $10US... The full
duplex speakerphones in these phones are as or better than a Polycom.   I'll
be happy to test this module -- I'm just not a C/++ guy.

Gerry
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From mike at jerris.com  Wed Mar  4 19:55:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 4 Mar 2009 22:55:59 -0500
Subject: [Freeswitch-users] mod_unistim?
In-Reply-To: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com>
References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com>
Message-ID: <BCF277AD-E4FE-43D1-9AC0-343E501AD55A@jerris.com>

Due to licensing reasons, you can not "port" a gpl piece of code to  
FreeSWITCH due to restrictions imposed by the gpl so it is not  
possible to do this unless all copy-write holders approve a license  
change.

Mike

On Mar 4, 2009, at 10:31 PM, Gerry Hull wrote:

> I hear rumors that someone is porting chan_unistim to mod_unistim  
> for FreeSwitch??  I hope so -- I use this on my Asterisk box and  
> would love to use it with FS.   There are TONS of i2004 phones on  
> the surplus market these days... I've been buying NOS i2004s,  
> virgin, for less than $10US... The full duplex speakerphones in  
> these phones are as or better than a Polycom.   I'll be happy to  
> test this module -- I'm just not a C/++ guy.
>
> Gerry
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 brian at freeswitch.org  Wed Mar  4 20:00:40 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 Mar 2009 22:00:40 -0600
Subject: [Freeswitch-users] mod_unistim?
In-Reply-To: <BCF277AD-E4FE-43D1-9AC0-343E501AD55A@jerris.com>
References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com>
	<BCF277AD-E4FE-43D1-9AC0-343E501AD55A@jerris.com>
Message-ID: <FD21357D-2D1E-4EE7-B4C2-72B90733EF4E@freeswitch.org>

Actually in this case you can we were giving FULL rights to do what we  
wanted with the code from the original author.  ;)  I still have the  
emails about it.. and someone asked me about this a few weeks ago.

/b

On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote:

> Due to licensing reasons, you can not "port" a gpl piece of code to
> FreeSWITCH due to restrictions imposed by the gpl so it is not
> possible to do this unless all copy-write holders approve a license
> change.
>
> Mike



From kawarod at laposte.net  Thu Mar  5 04:12:51 2009
From: kawarod at laposte.net (rod)
Date: Thu, 05 Mar 2009 16:12:51 +0400
Subject: [Freeswitch-users] Rewriting Remote Party ID
Message-ID: <49AFC1C3.9030603@laposte.net>

Dear list,

I'd like to rewrite the number in the Remote Party ID header and only in 
this header.

ex: I'd like to prefix the caller  ID with a prefix code (000 in this 
example) in the  RPID header :

        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
        Remote-Party-ID: 
<sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling

should become:
        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
        Remote-Party-ID: 
<sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling

But the From field should remain unchanged.

And how to strip this prefix:
        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
        Remote-Party-ID: 
<sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling

should become:
        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
        Remote-Party-ID: 
<sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling


regards.


From brian at freeswitch.org  Thu Mar  5 04:23:02 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 06:23:02 -0600
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <49AFC1C3.9030603@laposte.net>
References: <49AFC1C3.9030603@laposte.net>
Message-ID: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>

Well this depends on how you're placing the call.. if its a standard  
bridge you can on the A-Leg set "effective_caller_id_number=000$ 
{caller_id_number}" before you call bridge.

Is the from already in the correct format?

/b

On Mar 5, 2009, at 6:12 AM, rod wrote:

> Dear list,
>
> I'd like to rewrite the number in the Remote Party ID header and  
> only in
> this header.
>
> ex: I'd like to prefix the caller  ID with a prefix code (000 in this
> example) in the  RPID header :
>
>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>        Remote-Party-ID:
> <sip: 
> 123456 
> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>
> should become:
>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>        Remote-Party-ID:
> <sip: 
> 000123456 
> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>
> But the From field should remain unchanged.
>
> And how to strip this prefix:
>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>        Remote-Party-ID:
> <sip: 
> 000123456 
> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>
> should become:
>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>        Remote-Party-ID:
> <sip: 
> 123456 
> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>
>
> regards.

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From kawarod at laposte.net  Thu Mar  5 04:46:51 2009
From: kawarod at laposte.net (rod)
Date: Thu, 05 Mar 2009 16:46:51 +0400
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>
References: <49AFC1C3.9030603@laposte.net>
	<165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>
Message-ID: <49AFC9BB.9090106@laposte.net>

Hi Brian,

if I use the function effective_caller_id_number with my INVITE, I get this:

From: "Anonymous" <sip:000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS.

this is not exactly what I'm looking for :p

rod


Brian West wrote:
> Well this depends on how you're placing the call.. if its a standard 
> bridge you can on the A-Leg set 
> "effective_caller_id_number=000${caller_id_number}" before you call 
> bridge.
>
> Is the from already in the correct format?
>
> /b
>
> On Mar 5, 2009, at 6:12 AM, rod wrote:
>
>> Dear list,
>>
>> I'd like to rewrite the number in the Remote Party ID header and only in 
>> this header.
>>
>> ex: I'd like to prefix the caller  ID with a prefix code (000 in this 
>> example) in the  RPID header :
>>
>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>        Remote-Party-ID: 
>> <sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>
>> should become:
>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>        Remote-Party-ID: 
>> <sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>
>> But the From field should remain unchanged.
>>
>> And how to strip this prefix:
>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>        Remote-Party-ID: 
>> <sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>
>> should become:
>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>        Remote-Party-ID: 
>> <sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>
>>
>> regards.
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 kawarod at laposte.net  Thu Mar  5 05:00:50 2009
From: kawarod at laposte.net (rod)
Date: Thu, 05 Mar 2009 17:00:50 +0400
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <49AFC9BB.9090106@laposte.net>
References: <49AFC1C3.9030603@laposte.net>	<165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>
	<49AFC9BB.9090106@laposte.net>
Message-ID: <49AFCD02.2000603@laposte.net>

the A leg invite looks like this:
From: "Anonymous"<sip:anonymous at anonymous.invalid;user=phone>

it has been rewritten like this:
From: "Anonymous" <sip:000000anonymous at 172.29.0.5>

rod

rod wrote:
> Hi Brian,
>
> if I use the function effective_caller_id_number with my INVITE, I get this:
>
> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS.
>
> this is not exactly what I'm looking for :p
>
> rod
>
>
> Brian West wrote:
>   
>> Well this depends on how you're placing the call.. if its a standard 
>> bridge you can on the A-Leg set 
>> "effective_caller_id_number=000${caller_id_number}" before you call 
>> bridge.
>>
>> Is the from already in the correct format?
>>
>> /b
>>
>> On Mar 5, 2009, at 6:12 AM, rod wrote:
>>
>>     
>>> Dear list,
>>>
>>> I'd like to rewrite the number in the Remote Party ID header and only in 
>>> this header.
>>>
>>> ex: I'd like to prefix the caller  ID with a prefix code (000 in this 
>>> example) in the  RPID header :
>>>
>>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>        Remote-Party-ID: 
>>> <sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>
>>> should become:
>>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>        Remote-Party-ID: 
>>> <sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>
>>> But the From field should remain unchanged.
>>>
>>> And how to strip this prefix:
>>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>        Remote-Party-ID: 
>>> <sip:000123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>
>>> should become:
>>>        From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>        Remote-Party-ID: 
>>> <sip:123456 at 10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>
>>>
>>> regards.
>>>       
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 gibbedhead at gmail.com  Thu Mar  5 02:14:57 2009
From: gibbedhead at gmail.com (J Mann/Harry)
Date: Thu, 5 Mar 2009 05:14:57 -0500
Subject: [Freeswitch-users] Please end the torment
Message-ID: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>

No, I've yet to contribute anything, I barely have my system doing
what I want. But I REALLY love Freeswitch and I want to see it BURY
Asterisk. (Windows server user here)

I've been struggling with the XML configs, trying to figure out what
does what and where! That's fine, I'm used to it. What I'm NOT used to
is the total lack of a forum-based community to join and participate
in! Where can users SHARE their configs, help each other, learn from
each others mistakes? No DEV forum? I'm speechless.

Let's be serious guys, mailing lists went out in the 90s! IRC is "meh"
at best! I want stickies, a forum for noobs, converts, a dev forum...
So on...

"WELCOME To FreeSWITCH!"

Am I asking too much here? A FORUM?

I can't see how you can spark interest when we're so sorely lacking
the most basic and widely used community environments on the net!

HELP SOMEBODY? Install  SMF ASAP!
http://www.simplemachines.org

BTW: I hate people who voice problems without offering viable
solutions in the process... Disgusting! So if someone can offer up a
simple hosting account, Control Panel 10, Windows Linux whatever...
I'll be more than happy to have SMF setup and receiving user
registrations within 24 hours! I've done it dozens of times before. I
will then gladly turn over the keys to the kingdom to the "powers that
be" and take a backseat, being simply a happy user from that point on!

Please folks, please. I'm dying over here and I'm sure I'm not alone!
I'm searching Google and finding nothing!! FORUMS!

Harry (email me here)
switchserver at gmail.com (my FS email)


From mrene_lists at avgs.ca  Thu Mar  5 06:04:15 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 5 Mar 2009 09:04:15 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
Message-ID: <A089AAB9-FA3D-4F82-98B9-D35193710948@avgs.ca>

The wiki explains all that and allows all that, whats wrong with IRC?
Come and ask questions if you don't understand, you'll get your  
answers quicker than complaining about the lack of forum.

Math

On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote:

> No, I've yet to contribute anything, I barely have my system doing
> what I want. But I REALLY love Freeswitch and I want to see it BURY
> Asterisk. (Windows server user here)
>
> I've been struggling with the XML configs, trying to figure out what
> does what and where! That's fine, I'm used to it. What I'm NOT used to
> is the total lack of a forum-based community to join and participate
> in! Where can users SHARE their configs, help each other, learn from
> each others mistakes? No DEV forum? I'm speechless.
>
> Let's be serious guys, mailing lists went out in the 90s! IRC is "meh"
> at best! I want stickies, a forum for noobs, converts, a dev forum...
> So on...
>
> "WELCOME To FreeSWITCH!"
>
> Am I asking too much here? A FORUM?
>
> I can't see how you can spark interest when we're so sorely lacking
> the most basic and widely used community environments on the net!
>
> HELP SOMEBODY? Install  SMF ASAP!
> http://www.simplemachines.org
>
> BTW: I hate people who voice problems without offering viable
> solutions in the process... Disgusting! So if someone can offer up a
> simple hosting account, Control Panel 10, Windows Linux whatever...
> I'll be more than happy to have SMF setup and receiving user
> registrations within 24 hours! I've done it dozens of times before. I
> will then gladly turn over the keys to the kingdom to the "powers that
> be" and take a backseat, being simply a happy user from that point on!
>
> Please folks, please. I'm dying over here and I'm sure I'm not alone!
> I'm searching Google and finding nothing!! FORUMS!
>
> Harry (email me here)
> switchserver at gmail.com (my FS email)
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 suspicious.org  Thu Mar  5 06:11:19 2009
From: krice at suspicious.org (Ken Rice)
Date: Thu, 05 Mar 2009 08:11:19 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <A089AAB9-FA3D-4F82-98B9-D35193710948@avgs.ca>
Message-ID: <C5D539A7.4FCB4%krice@suspicious.org>

Not to mention there is a full archive of the mailing list on nabble

http://www.nabble.com/Freeswitch-users-f32209.html



> From: Mathieu Rene <mrene_lists at avgs.ca>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 5 Mar 2009 09:04:15 -0500
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Please end the torment
> 
> The wiki explains all that and allows all that, whats wrong with IRC?
> Come and ask questions if you don't understand, you'll get your
> answers quicker than complaining about the lack of forum.
> 
> Math
> 
> On 5-Mar-09, at 5:14 AM, J Mann/Harry wrote:
> 
>> No, I've yet to contribute anything, I barely have my system doing
>> what I want. But I REALLY love Freeswitch and I want to see it BURY
>> Asterisk. (Windows server user here)
>> 
>> I've been struggling with the XML configs, trying to figure out what
>> does what and where! That's fine, I'm used to it. What I'm NOT used to
>> is the total lack of a forum-based community to join and participate
>> in! Where can users SHARE their configs, help each other, learn from
>> each others mistakes? No DEV forum? I'm speechless.
>> 
>> Let's be serious guys, mailing lists went out in the 90s! IRC is "meh"
>> at best! I want stickies, a forum for noobs, converts, a dev forum...
>> So on...
>> 
>> "WELCOME To FreeSWITCH!"
>> 
>> Am I asking too much here? A FORUM?
>> 
>> I can't see how you can spark interest when we're so sorely lacking
>> the most basic and widely used community environments on the net!
>> 
>> HELP SOMEBODY? Install  SMF ASAP!
>> http://www.simplemachines.org
>> 
>> BTW: I hate people who voice problems without offering viable
>> solutions in the process... Disgusting! So if someone can offer up a
>> simple hosting account, Control Panel 10, Windows Linux whatever...
>> I'll be more than happy to have SMF setup and receiving user
>> registrations within 24 hours! I've done it dozens of times before. I
>> will then gladly turn over the keys to the kingdom to the "powers that
>> be" and take a backseat, being simply a happy user from that point on!
>> 
>> Please folks, please. I'm dying over here and I'm sure I'm not alone!
>> I'm searching Google and finding nothing!! FORUMS!
>> 
>> Harry (email me here)
>> switchserver at gmail.com (my FS email)
>> 
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mike at jerris.com  Thu Mar  5 06:22:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2009 06:22:32 -0800 (PST)
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <C5D539A7.4FCB4%krice@suspicious.org>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<A089AAB9-FA3D-4F82-98B9-D35193710948@avgs.ca>
	<C5D539A7.4FCB4%krice@suspicious.org>
Message-ID: <1236262952915-2429661.post@n2.nabble.com>


Much more than an archive, nabble makes a full embeddable forum that is
linked to the mailing list.  We will be embedding this in our webpage soon
for the best of both worlds, a forum and a mailing list without the
additional overhead of having to monitor 2 things.

Mike


Ken Rice-3 wrote:
> 
> Not to mention there is a full archive of the mailing list on nabble
> 
> http://www.nabble.com/Freeswitch-users-f32209.html
> 
> 

-- 
View this message in context: http://n2.nabble.com/Please-end-the-torment-tp2429589p2429661.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From damin at nacs.net  Thu Mar  5 06:46:44 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Thu, 5 Mar 2009 09:46:44 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <C5D539A7.4FCB4%krice@suspicious.org>
References: <A089AAB9-FA3D-4F82-98B9-D35193710948@avgs.ca>
	<C5D539A7.4FCB4%krice@suspicious.org>
Message-ID: <123301c99da1$33ee98e0$9bcbcaa0$@net>

I hate forums. Forums suck. They are a pain in the ass to search and find
things. I prefer the mailing list and the Wiki. Can we please keep it that
way? It is really easy to find stuff via Nabble and the Wiki.



From gmaruzz at celliax.org  Thu Mar  5 07:11:25 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 5 Mar 2009 16:11:25 +0100
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <1236262952915-2429661.post@n2.nabble.com>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<A089AAB9-FA3D-4F82-98B9-D35193710948@avgs.ca>
	<C5D539A7.4FCB4%krice@suspicious.org>
	<1236262952915-2429661.post@n2.nabble.com>
Message-ID: <7b197bef0903050711v5a8d0eefo31c9e476fe6167ae@mail.gmail.com>

On Thu, Mar 5, 2009 at 3:22 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Much more than an archive, nabble makes a full embeddable forum that is
> linked to the mailing list. ?We will be embedding this in our webpage soon
> for the best of both worlds, a forum and a mailing list without the
> additional overhead of having to monitor 2 things.

agree!


From helmut.kuper at ewetel.de  Thu Mar  5 07:31:34 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 05 Mar 2009 16:31:34 +0100
Subject: [Freeswitch-users] Patch for mod_event_socket to fire real CUSTOM
 events via sendevent command
Message-ID: <49AFF056.70706@ewetel.de>

Hello,

I enhanced mod_event_socket's sendevent command to fire real CUSTOM
events with correct subclass and trailing custom headers, so that
subscribed nodes can receive those events. Is there a way to get the
patch into trunk?

regards
Helmut



From csorlie at teldio.com  Thu Mar  5 07:41:28 2009
From: csorlie at teldio.com (Cameron Sorlie)
Date: Thu, 5 Mar 2009 10:41:28 -0500
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
Message-ID: <d8d3ef6d0903050741p32859c6ds72dd58219d201bf@mail.gmail.com>

In a sense, you might say I did futz with mod_vmd ... to create mod_vad.
There appeared to be just no (easy) way to modify the internal VAD code in
the FreeSWITCH core (see switch_rtp.c) to identify the origins of voice
activity. And rather than build into mod_vmd, which is a special purpose
tool, a separate module for VAD seemed like a reasonable idea.

In short, the mod_vad which I've written up independently monitors the read
and the write legs of the session it is run on, and tags each VAD_TALK and
VAD_NOTALK event it fires with a user-supplied identification marker (a
short string) for the leg which the event relates to. At the moment, the VAD
algorithm is dead simple, and is much like the one in the core. I will be
happy to submit this module, in a little while, after I've had a chance to
make it perhaps a bit more useable outside of our own application context.

Cam

On Mon, Mar 2, 2009 at 5:43 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

>
> On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > i think that's what mod_vmd does
> >
> I think that's right. It just does the opposite - instead of looking
> for differing power levels it looks for the same power level. In other
> words it tries to detect distinctly non-human sound. I'll bet you
> could futz with that code and have it fire off events when it detects
> what it believes is human speech.
>
> -MC
>
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From mrene_lists at avgs.ca  Thu Mar  5 07:41:33 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 5 Mar 2009 10:41:33 -0500
Subject: [Freeswitch-users] Patch for mod_event_socket to fire real
	CUSTOM events via sendevent command
In-Reply-To: <49AFF056.70706@ewetel.de>
References: <49AFF056.70706@ewetel.de>
Message-ID: <8BC75FF7-C82D-49E4-BFEA-E40887372A57@avgs.ca>

Sure, open a JIRA as a improvement, and prefix your bug name with  
[patch]

http://jira.freeswitch.org/

Math

On 5-Mar-09, at 10:31 AM, Helmut Kuper wrote:

> Hello,
>
> I enhanced mod_event_socket's sendevent command to fire real CUSTOM
> events with correct subclass and trailing custom headers, so that
> subscribed nodes can receive those events. Is there a way to get the
> patch into trunk?
>
> regards
> Helmut
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 brian at freeswitch.org  Thu Mar  5 07:51:15 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 09:51:15 -0600
Subject: [Freeswitch-users] Detecting the origin of voice activity using
	VAD
In-Reply-To: <d8d3ef6d0903050741p32859c6ds72dd58219d201bf@mail.gmail.com>
References: <d8d3ef6d0903050741p32859c6ds72dd58219d201bf@mail.gmail.com>
Message-ID: <9AE4AAC4-34A3-4758-B64E-38B0056C5F30@freeswitch.org>

Kewl you can contribute via http://jira.freeswitch.org  ;)

/b

On Mar 5, 2009, at 9:41 AM, Cameron Sorlie wrote:

> In a sense, you might say I did futz with mod_vmd ... to create  
> mod_vad. There appeared to be just no (easy) way to modify the  
> internal VAD code in the FreeSWITCH core (see switch_rtp.c) to  
> identify the origins of voice activity. And rather than build into  
> mod_vmd, which is a special purpose tool, a separate module for VAD  
> seemed like a reasonable idea.
>
> In short, the mod_vad which I've written up independently monitors  
> the read and the write legs of the session it is run on, and tags  
> each VAD_TALK and VAD_NOTALK event it fires with a user-supplied  
> identification marker (a short string) for the leg which the event  
> relates to. At the moment, the VAD algorithm is dead simple, and is  
> much like the one in the core. I will be happy to submit this  
> module, in a little while, after I've had a chance to make it  
> perhaps a bit more useable outside of our own application context.
>
> Cam



From helmut.kuper at ewetel.de  Thu Mar  5 07:52:27 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 05 Mar 2009 16:52:27 +0100
Subject: [Freeswitch-users] Problems with record_stereo
In-Reply-To: <D719BF4A-9D11-4C29-B06A-922DC89C6FC7@freeswitch.org>
References: <49A7F393.6080406@ewetel.de>	<87f2f3b90903021005y47d31796l5924f8f042b6cf3c@mail.gmail.com>	<49ACE742.5090809@ewetel.de>	<A0EBCA0F-110F-4CEF-A735-61D90CE7E9D5@freeswitch.org>	<49AD4BAA.8090208@ewetel.de>	<9D3C9BB6-4F69-4975-972F-30EC7C76EDAA@freeswitch.org>	<49AE7404.8000905@ewetel.de>	<49AE7D43.9070409@ewetel.de>	<3DC88EF0-DFD8-446A-92CB-DEC33F15BAB9@freeswitch.org>	<49AEA042.3090908@ewetel.de>
	<D719BF4A-9D11-4C29-B06A-922DC89C6FC7@freeswitch.org>
Message-ID: <49AFF53B.5000802@ewetel.de>

Hello Brian,

I checked the files without earphones and with Audacity. You are right
Brian, everything is fine. Sorry for the inconvenience.

Thx for your patience.

regards
helmut




From BenHoltsclaw at averyschools.net  Thu Mar  5 09:03:50 2009
From: BenHoltsclaw at averyschools.net (Ben Holtsclaw)
Date: Thu, 05 Mar 2009 12:03:50 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
Message-ID: <49AFBFA6.45B7.0079.0@averyschools.net>

I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this:
 
[23:10] <{tasker}> me, too, but i'm a different animal
[23:10] <{tasker}> in NY and in Miami i went nutz
[23:10] <diegoviola> lol
[23:10] * jefferai is now known as lollerai
[23:10] <diegoviola> yeah i love her
[23:10] <{tasker}> latinas everywhere
[23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch
[23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out) )
[23:11] <hmmhesays> here its blond blue eyed girls
[23:11] * lollerai is now known as lolferai
[23:11] <rpm> brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there
[23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days"
[23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
[23:11] <{tasker}> blonde / blue eyes are overrated
[23:11] <{tasker}> give me a latina any day
[23:11] <hmmhesays> best thing around here though

If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation??
 
ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers.
 
 
Ben Holtsclaw
Network Engineer
Avery County Schools
Ph:  828.733.3567 x2301

>>> On 3/5/2009 at 5:14 AM, "J Mann/Harry" <gibbedhead at gmail.com> wrote:
No, I've yet to contribute anything, I barely have my system doing
what I want. But I REALLY love Freeswitch and I want to see it BURY
Asterisk. (Windows server user here)

I've been struggling with the XML configs, trying to figure out what
does what and where! That's fine, I'm used to it. What I'm NOT used to
is the total lack of a forum-based community to join and participate
in! Where can users SHARE their configs, help each other, learn from
each others mistakes? No DEV forum? I'm speechless.

Let's be serious guys, mailing lists went out in the 90s! IRC is "meh"
at best! I want stickies, a forum for noobs, converts, a dev forum...
So on...

"WELCOME To FreeSWITCH!"

Am I asking too much here? A FORUM?

I can't see how you can spark interest when we're so sorely lacking
the most basic and widely used community environments on the net!

HELP SOMEBODY? Install  SMF ASAP!
http://www.simplemachines.org 

BTW: I hate people who voice problems without offering viable
solutions in the process... Disgusting! So if someone can offer up a
simple hosting account, Control Panel 10, Windows Linux whatever...
I'll be more than happy to have SMF setup and receiving user
registrations within 24 hours! I've done it dozens of times before. I
will then gladly turn over the keys to the kingdom to the "powers that
be" and take a backseat, being simply a happy user from that point on!

Please folks, please. I'm dying over here and I'm sure I'm not alone!
I'm searching Google and finding nothing!! FORUMS!

Harry (email me here)
switchserver at gmail.com (my FS email)

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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  Thu Mar  5 09:16:08 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 11:16:08 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
Message-ID: <CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>

I have been trying to push all the social talk into #freeswitch-social  
to keep #freeswitch on topic.. sometimes after hours in the US it gets  
a bit off topic.  I'm usually alive in the channel till around 11PM+  
CST most days.  I take questions and answer questions at all hours if  
I'm awake... I too am guilty of going off topic.

/b


On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:

> I agree with Harry. I do not like the mailing list. Those that do  
> like the mailing list always advocate Nabble. For those that  
> advocate that solution, do you even realize that you can't post on  
> Nabble unless you are subscribed to the mailing list? I am also not  
> a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in  
> years! I think I uninstalled my IRC client at the same time I  
> uninstalled my NNTP reader. Most of the time, I actually find it  
> difficult to obtain support in the #freeswitch channel. Once you ask  
> the question, if somebody doesn't happen to be there that knows the  
> answer, then you're screwed. How many times have I asked a question  
> only to wait 30 seconds and then see, "anthm has joined  
> #freeswitch." Crap...do I ask the question again? I have found the  
> conversation in #openzap to be much more focused. Thank goodness I'm  
> using that module! In that channel, I never see conversations about  
> cd burners, somebody's girlfriend in South America, or off color  
> jokes about someone's sexual proclivity. And because I know I'll get  
> flamed for saying that, just look at this:
>
> [23:10] <{tasker}> me, too, but i'm a different animal
> [23:10] <{tasker}> in NY and in Miami i went nutz
> [23:10] <diegoviola> lol
> [23:10] * jefferai is now known as lollerai
> [23:10] <diegoviola> yeah i love her
> [23:10] <{tasker}> latinas everywhere
> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left  
> #freeswitch
> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection  
> timed out))
> [23:11] <hmmhesays> here its blond blue eyed girls
> [23:11] * lollerai is now known as lolferai
> [23:11] <rpm> brazilians... hopefully she's hot. i've seen some  
> pretty dodgy looking chicks from there
> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her,  
> "yeaaaaaaaaah, it's just for a few days"
> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
> [23:11] <{tasker}> blonde / blue eyes are overrated
> [23:11] <{tasker}> give me a latina any day
> [23:11] <hmmhesays> best thing around here though
>
> If I'm going into #freeswitch at 11pm at night, it's probably  
> because I really need some help with some problem I've run into  
> after hours. Can you imagine me injecting a question about a SIP  
> profile into that conversation??
>
> ALL that aside... I'm willing to use a carrier pigeon if that's the  
> way the three primary developers wish to communicate. They have been  
> instrumental in getting my project where it is today. You know the  
> saying... beggars can't be choosers.
>
>
> Ben Holtsclaw
> Network Engineer
> Avery County Schools
> Ph:  828.733.3567 x2301
>

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From mrene_lists at avgs.ca  Thu Mar  5 09:19:31 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 5 Mar 2009 12:19:31 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
Message-ID: <CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>

The bot even takes people's question and keeps them in list so when  
someone that knows shows up he can answer it, thats the ~take-a-number  
option...

On 5-Mar-09, at 12:16 PM, Brian West wrote:

> I have been trying to push all the social talk into #freeswitch- 
> social to keep #freeswitch on topic.. sometimes after hours in the  
> US it gets a bit off topic.  I'm usually alive in the channel till  
> around 11PM+ CST most days.  I take questions and answer questions  
> at all hours if I'm awake... I too am guilty of going off topic.
>
> /b
>
>
> On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:
>
>> I agree with Harry. I do not like the mailing list. Those that do  
>> like the mailing list always advocate Nabble. For those that  
>> advocate that solution, do you even realize that you can't post on  
>> Nabble unless you are subscribed to the mailing list? I am also not  
>> a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in  
>> years! I think I uninstalled my IRC client at the same time I  
>> uninstalled my NNTP reader. Most of the time, I actually find it  
>> difficult to obtain support in the #freeswitch channel. Once you  
>> ask the question, if somebody doesn't happen to be there that knows  
>> the answer, then you're screwed. How many times have I asked a  
>> question only to wait 30 seconds and then see, "anthm has joined  
>> #freeswitch." Crap...do I ask the question again? I have found the  
>> conversation in #openzap to be much more focused. Thank goodness  
>> I'm using that module! In that channel, I never see conversations  
>> about cd burners, somebody's girlfriend in South America, or off  
>> color jokes about someone's sexual proclivity. And because I know  
>> I'll get flamed for saying that, just look at this:
>>
>> [23:10] <{tasker}> me, too, but i'm a different animal
>> [23:10] <{tasker}> in NY and in Miami i went nutz
>> [23:10] <diegoviola> lol
>> [23:10] * jefferai is now known as lollerai
>> [23:10] <diegoviola> yeah i love her
>> [23:10] <{tasker}> latinas everywhere
>> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has  
>> left #freeswitch
>> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection  
>> timed out))
>> [23:11] <hmmhesays> here its blond blue eyed girls
>> [23:11] * lollerai is now known as lolferai
>> [23:11] <rpm> brazilians... hopefully she's hot. i've seen some  
>> pretty dodgy looking chicks from there
>> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her,  
>> "yeaaaaaaaaah, it's just for a few days"
>> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
>> [23:11] <{tasker}> blonde / blue eyes are overrated
>> [23:11] <{tasker}> give me a latina any day
>> [23:11] <hmmhesays> best thing around here though
>>
>> If I'm going into #freeswitch at 11pm at night, it's probably  
>> because I really need some help with some problem I've run into  
>> after hours. Can you imagine me injecting a question about a SIP  
>> profile into that conversation??
>>
>> ALL that aside... I'm willing to use a carrier pigeon if that's the  
>> way the three primary developers wish to communicate. They have  
>> been instrumental in getting my project where it is today. You know  
>> the saying... beggars can't be choosers.
>>
>>
>> Ben Holtsclaw
>> Network Engineer
>> Avery County Schools
>> Ph:  828.733.3567 x2301
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 egghunt at gmail.com  Thu Mar  5 09:27:59 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Thu, 5 Mar 2009 14:27:59 -0300
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
Message-ID: <ff2539090903050927g56470165xaca8e1d770c3987e@mail.gmail.com>

Besides #freeswitch-social and #openzap, there are other channels related to
the project. List:
http://wiki.freeswitch.org/wiki/Main_Page#Community_and_Support

On Thu, Mar 5, 2009 at 2:19 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The bot even takes people's question and keeps them in list so when someone
> that knows shows up he can answer it, thats the ~take-a-number option...
> On 5-Mar-09, at 12:16 PM, Brian West wrote:
>
> I have been trying to push all the social talk into #freeswitch-social to
> keep #freeswitch on topic.. sometimes after hours in the US it gets a bit
> off topic.  I'm usually alive in the channel till around 11PM+ CST most
> days.  I take questions and answer questions at all hours if I'm awake... I
> too am guilty of going off topic.
> /b
>
>
> On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:
>
> I agree with Harry. I do not like the mailing list. Those that do like the
> mailing list always advocate Nabble. For those that advocate that solution,
> do you even realize that you can't post on Nabble unless you *are* subscribed
> to the mailing list? I am also not a fan of IRC. Before I came upon
> FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client
> at the same time I uninstalled my NNTP reader. Most of the time, I actually
> find it difficult to obtain support in the #freeswitch channel. Once you ask
> the question, if somebody doesn't happen to be there that knows the answer,
> then you're screwed. How many times have I asked a question only to wait 30
> seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the
> question again? I *have* found the conversation in #openzap to be much
> more focused. Thank goodness I'm using that module! In that channel, I never
> see conversations about cd burners, somebody's girlfriend in South America,
> or off color jokes about someone's sexual proclivity. And because I know
> I'll get flamed for saying that, just look at this:
>
> [23:10] <{tasker}> me, too, but i'm a different animal
> [23:10] <{tasker}> in NY and in Miami i went nutz
> [23:10] <diegoviola> lol
> [23:10] * jefferai is now known as lollerai
> [23:10] <diegoviola> yeah i love her
> [23:10] <{tasker}> latinas everywhere
> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left
> #freeswitch
> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed
> out) )
> [23:11] <hmmhesays> here its blond blue eyed girls
> [23:11] * lollerai is now known as lolferai
> [23:11] <rpm> brazilians... hopefully she's hot. i've seen some pretty
> dodgy looking chicks from there
> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her,
> "yeaaaaaaaaah, it's just for a few days"
> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
> [23:11] <{tasker}> blonde / blue eyes are overrated
> [23:11] <{tasker}> give me a latina any day
> [23:11] <hmmhesays> best thing around here though
>
> If I'm going into #freeswitch at 11pm at night, it's probably because I
> really need some help with some problem I've run into after hours. Can you
> imagine me injecting a question about a SIP profile into that conversation??
>
> ALL that aside... I'm willing to use a carrier pigeon if that's the way the
> three primary developers wish to communicate. They have been instrumental in
> getting my project where it is today. You know the saying... beggars can't
> be choosers.
>
>
> Ben Holtsclaw
> Network Engineer
> Avery County Schools
> Ph:  828.733.3567 x2301
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Arnaldo M Pereira
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From BenHoltsclaw at averyschools.net  Thu Mar  5 09:27:35 2009
From: BenHoltsclaw at averyschools.net (Ben Holtsclaw)
Date: Thu, 05 Mar 2009 12:27:35 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
Message-ID: <49AFC537.45B7.0079.0@averyschools.net>

The problem with take-a-number is what if I'm not there when someone can answer it?

>>> On 3/5/2009 at 12:19 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
The bot even takes people's question and keeps them in list so when someone that knows shows up he can answer it, thats the ~take-a-number option...

On 5-Mar-09, at 12:16 PM, Brian West wrote:



I have been trying to push all the social talk into #freeswitch-social to keep #freeswitch on topic.. sometimes after hours in the US it gets a bit off topic.  I'm usually alive in the channel till around 11PM+ CST most days.  I take questions and answer questions at all hours if I'm awake... I too am guilty of going off topic.  

/b


On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:




I agree with Harry. I do not like the mailing list. Those that do like the mailing list always advocate Nabble. For those that advocate that solution, do you even realize that you can't post on Nabble unless you are subscribed to the mailing list? I am also not a fan of IRC. Before I came upon FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client at the same time I uninstalled my NNTP reader. Most of the time, I actually find it difficult to obtain support in the #freeswitch channel. Once you ask the question, if somebody doesn't happen to be there that knows the answer, then you're screwed. How many times have I asked a question only to wait 30 seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the question again? I have found the conversation in #openzap to be much more focused. Thank goodness I'm using that module! In that channel, I never see conversations about cd burners, somebody's girlfriend in South America, or off color jokes about someone's sexual proclivity. And because I know I'll get flamed for saying that, just look at this:

 
[23:10] <{tasker}> me, too, but i'm a different animal
[23:10] <{tasker}> in NY and in Miami i went nutz
[23:10] <diegoviola> lol
[23:10] * jefferai is now known as lollerai
[23:10] <diegoviola> yeah i love her
[23:10] <{tasker}> latinas everywhere
[23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left #freeswitch
[23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed out))
[23:11] <hmmhesays> here its blond blue eyed girls
[23:11] * lollerai is now known as lolferai
[23:11] <rpm> brazilians... hopefully she's hot. i've seen some pretty dodgy looking chicks from there
[23:11] <{tasker}> diego: go back, buy her a ticket and tell her, "yeaaaaaaaaah, it's just for a few days"
[23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
[23:11] <{tasker}> blonde / blue eyes are overrated
[23:11] <{tasker}> give me a latina any day
[23:11] <hmmhesays> best thing around here though

If I'm going into #freeswitch at 11pm at night, it's probably because I really need some help with some problem I've run into after hours. Can you imagine me injecting a question about a SIP profile into that conversation??
 
ALL that aside... I'm willing to use a carrier pigeon if that's the way the three primary developers wish to communicate. They have been instrumental in getting my project where it is today. You know the saying... beggars can't be choosers.
 
 
Ben Holtsclaw
Network Engineer
Avery County Schools
Ph:  828.733.3567 x2301



_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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  Thu Mar  5 09:30:21 2009
From: gerry at pstn2.net (Gerry Hull)
Date: Thu, 5 Mar 2009 12:30:21 -0500
Subject: [Freeswitch-users] mod_unistim?
In-Reply-To: <FD21357D-2D1E-4EE7-B4C2-72B90733EF4E@freeswitch.org>
References: <98a86adf0903041931m4b431cean4b7173e08cea23e1@mail.gmail.com>
	<BCF277AD-E4FE-43D1-9AC0-343E501AD55A@jerris.com>
	<FD21357D-2D1E-4EE7-B4C2-72B90733EF4E@freeswitch.org>
Message-ID: <98a86adf0903050930t7b0df79fm5524a3ec0aaa3b4f@mail.gmail.com>

Yeah, I was looking at the Developers IRC log...  I see that milkj has given
you an MPL
license...  Has anyone started development on the port?

On Wed, Mar 4, 2009 at 11:00 PM, Brian West <brian at freeswitch.org> wrote:

> Actually in this case you can we were giving FULL rights to do what we
> wanted with the code from the original author.  ;)  I still have the
> emails about it.. and someone asked me about this a few weeks ago.
>
> /b
>
> On Mar 4, 2009, at 9:55 PM, Michael Jerris wrote:
>
> > Due to licensing reasons, you can not "port" a gpl piece of code to
> > FreeSWITCH due to restrictions imposed by the gpl so it is not
> > possible to do this unless all copy-write holders approve a license
> > change.
> >
> > Mike
>
>
>  _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mike at jerris.com  Thu Mar  5 09:31:14 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2009 12:31:14 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <49AFBFA6.45B7.0079.0@averyschools.net>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
Message-ID: <6F7B8F0B-0975-452D-A6C1-AE484131E2AB@jerris.com>


On Mar 5, 2009, at 12:03 PM, Ben Holtsclaw wrote:

> I agree with Harry. I do not like the mailing list. Those that do  
> like the mailing list always advocate Nabble. For those that  
> advocate that solution, do you even realize that you can't post on  
> Nabble unless you are subscribed to the mailing list?

Fair points, the plan to address this is that we are moving to a new  
back-end to our hosting infrastructure that has unified logins.  Once  
this is in place we can make it so you are a "member" of the mailing  
list as soon as you have an id, although set to not receive mail, then  
we'll have a subscriptions options page where you can just check the  
lists you want sent to you via email, and the nabble forums can be  
embedded in our page directly.  It is not perfect but much closer to  
something usable.  I understand that some are more comfortable than  
others with different modes of communication. We are trying to find a  
balance between being able to serve the community with the best way to  
communicate for them, and us being able to actually monitor and  
maintain it.  A forum is pretty useless if no one responds or monitors  
it and already we are flooded by the amount on the mailing list,  
having to answer the same question the same day in both because  
someone prefers one to the other and having to spend time checking  
both is not a good use of time.  Do you think that if we better  
integrate the nabble forums into our site and list subscriptions that  
it would be a usable system?

Mike

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From anthony.minessale at gmail.com  Thu Mar  5 09:33:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Mar 2009 11:33:03 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
Message-ID: <191c3a030903050933p6df4a725o9be751a390f9c916@mail.gmail.com>

Ok,

SO? We should make a forum?

See this one?

http://www.voip-info.org/boards/index.php?b=6

Yah, that one has been around for many months and nobody maintains it.  We
do have a link to it on our homepage but nobody reads it so, there is a
dilemma here right?

IRC is for sure more useful of a resource between 9am and 6pm GMT-6 in the
timezone where me and bkw live.  Any other times it would be up to other
community members like the few who have been stepping up in a major way in
the last few weeks, thank you all.

I don't know how good of a job we are doing, but we are a small group and we
already spend most of every day managing a bug tracker, irc, and this list.
If you took a poll 8/10 people hate forums.  I don't really know if i like
them or not, I have enjoyed finding forums before that discussed something
that happened ages ago that relates to a problem I have and being able to
skim months into the future reading the end results, but more times than not
they go south and contain a bunch of flaming back and forth about midway
through.

We already need voulenteers to help manage the resources we already have,
Website, Mailing list, JIRA.  If we introduce more we, run the risk of doing
a bad job at maintaining them.  Is anyone willing to help?

If you can use nabble to post as long as you have an account on the mailing
list, can't you just disable mail dilevery and use the nabble exclusively?
If not is there another solution?  I really need some help in this
department to even consider introducing any more community resources.






On Thu, Mar 5, 2009 at 11:16 AM, Brian West <brian at freeswitch.org> wrote:

> I have been trying to push all the social talk into #freeswitch-social to
> keep #freeswitch on topic.. sometimes after hours in the US it gets a bit
> off topic.  I'm usually alive in the channel till around 11PM+ CST most
> days.  I take questions and answer questions at all hours if I'm awake... I
> too am guilty of going off topic.
> /b
>
>
> On Mar 5, 2009, at 11:03 AM, Ben Holtsclaw wrote:
>
> I agree with Harry. I do not like the mailing list. Those that do like the
> mailing list always advocate Nabble. For those that advocate that solution,
> do you even realize that you can't post on Nabble unless you *are* subscribed
> to the mailing list? I am also not a fan of IRC. Before I came upon
> FreeSWITCH, I hadn't used IRC in years! I think I uninstalled my IRC client
> at the same time I uninstalled my NNTP reader. Most of the time, I actually
> find it difficult to obtain support in the #freeswitch channel. Once you ask
> the question, if somebody doesn't happen to be there that knows the answer,
> then you're screwed. How many times have I asked a question only to wait 30
> seconds and then see, "anthm has joined #freeswitch." Crap...do I ask the
> question again? I *have* found the conversation in #openzap to be much
> more focused. Thank goodness I'm using that module! In that channel, I never
> see conversations about cd burners, somebody's girlfriend in South America,
> or off color jokes about someone's sexual proclivity. And because I know
> I'll get flamed for saying that, just look at this:
>
> [23:10] <{tasker}> me, too, but i'm a different animal
> [23:10] <{tasker}> in NY and in Miami i went nutz
> [23:10] <diegoviola> lol
> [23:10] * jefferai is now known as lollerai
> [23:10] <diegoviola> yeah i love her
> [23:10] <{tasker}> latinas everywhere
> [23:11] * lanwifie (n=Sami at 99-196-39-200.cust.wildblue.net) has left
> #freeswitch
> [23:11] * jjg (n=jjg at 76.21.4.40) Quit (Read error: 110 (Connection timed
> out) )
> [23:11] <hmmhesays> here its blond blue eyed girls
> [23:11] * lollerai is now known as lolferai
> [23:11] <rpm> brazilians... hopefully she's hot. i've seen some pretty
> dodgy looking chicks from there
> [23:11] <{tasker}> diego: go back, buy her a ticket and tell her,
> "yeaaaaaaaaah, it's just for a few days"
> [23:11] * martyn-dev (n=martyn-d at 190.26.4.61) has joined #freeswitch
> [23:11] <{tasker}> blonde / blue eyes are overrated
> [23:11] <{tasker}> give me a latina any day
> [23:11] <hmmhesays> best thing around here though
>
> If I'm going into #freeswitch at 11pm at night, it's probably because I
> really need some help with some problem I've run into after hours. Can you
> imagine me injecting a question about a SIP profile into that conversation??
>
> ALL that aside... I'm willing to use a carrier pigeon if that's the way the
> three primary developers wish to communicate. They have been instrumental in
> getting my project where it is today. You know the saying... beggars can't
> be choosers.
>
>
> Ben Holtsclaw
> Network Engineer
> Avery County Schools
> Ph:  828.733.3567 x2301
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From brian at freeswitch.org  Thu Mar  5 09:37:36 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 11:37:36 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <49AFC537.45B7.0079.0@averyschools.net>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
	<49AFC537.45B7.0079.0@averyschools.net>
Message-ID: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>

Well you can always shoot an email to the mailing list... or wait for  
someone that can answer it.. which reminds me we really need people to  
volunteer to help out on the IRC channel so we have skilled people  
able to answer questions 24/7, I personally cover 12-16 hours a day  
most of the time.  Even weekends!  I was even up at 4am today and  
answered a few.

/b

On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote:

> The problem with take-a-number is what if I'm not there when someone  
> can answer it?



From egghunt at gmail.com  Thu Mar  5 09:43:04 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Thu, 5 Mar 2009 14:43:04 -0300
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
	<49AFC537.45B7.0079.0@averyschools.net>
	<600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
Message-ID: <ff2539090903050943k59598cfvb70830eca1fb4b18@mail.gmail.com>

On Thu, Mar 5, 2009 at 2:37 PM, Brian West <brian at freeswitch.org> wrote:

> Well you can always shoot an email to the mailing list... or wait for
> someone that can answer it.. which reminds me we really need people to
> volunteer to help out on the IRC channel so we have skilled people
> able to answer questions 24/7, I personally cover 12-16 hours a day
> most of the time.  Even weekends!  I was even up at 4am today and
> answered a few.


I'm usually on the channel starting 9am (GMT -3). Can't answer every
question like you guys, but I can surely help.


>
>
> /b
>
> On Mar 5, 2009, at 11:27 AM, Ben Holtsclaw wrote:
>
> > The problem with take-a-number is what if I'm not there when someone
> > can answer it?
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Arnaldo M Pereira
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From damin at nacs.net  Thu Mar  5 09:43:43 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Thu, 5 Mar 2009 12:43:43 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>	<49AFBFA6.45B7.0079.0@averyschools.net>	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>	<49AFC537.45B7.0079.0@averyschools.net>
	<600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
Message-ID: <131e01c99db9$ed440ba0$c7cc22e0$@net>

You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :)




From Prometheus001 at gmx.net  Thu Mar  5 12:20:18 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 05 Mar 2009 21:20:18 +0100
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>
References: <49AC6EE4.9080509@gmx.net>	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>	<49AD0F7B.7000802@gmx.net>	<57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>	<49ADD3E4.20408@gmx.net>
	<8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>
Message-ID: <49B03402.8050601@gmx.net>

Hello Brian,

concerning
> Well you should use ESL then ;)
I simply do not understand what you mean by this. Is it sarcastic? Am I
asking stupid questions?

After upgrading Freeswitch to the newest trunk, mod_pocketsphinx didn't
work anymore. So I asked this mailing list about information about what
happened. I understand now that there were some significant changes in
mod_pocketsphinx and that also some other files have to be updated. I
could not find any documentation about these changes, and asking here on
this mailing list was rather disappointing for me. Some bits, yes. Some
things don't work/crash, as I have read here.
We are not using Freeswitch just as a toy to play around. Sometimes it's
simply important to know which impact a certain change may have on our
system. And other people will run into the same problem. So any advice
was needed about the status and how to make it work. I'll update the
wiki with this information (as I usually do), I promise.

I honor the great work you do and freeswitch is really great. But asking:
>Will there also be major changes in the events I receive through
mod_eventsocket?
>Will I need updated grammar files for the other models too?

and receiving
> Well you should use ESL then ;)

is frustrating.


Best regards
Peter


Brian West schrieb:
> Well you should use ESL then ;)
>
> /b
>
> On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote:
>
>   
>> Thank you Brian,
>>
>> I will try this later.
>>
>> Currently I was happy to get this working on SVN 10003.
>>
>> As mod_pockesphinx has changed/evolved significantely: Will there also
>> be major changes in the events I receive through  mod_eventsocket?
>> I spend some time on parsing the right data out of the eventsocket
>> interface, and I would just have an idea, if I will have to expect
>> significant work to do, when I later switch to the current SVN.
>>
>> Will I need updated grammar files for the other models too?
>>
>> Best regards
>> Peter
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Thu Mar  5 12:32:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Mar 2009 12:32:29 -0800
Subject: [Freeswitch-users] pocketsphinx and event socket
In-Reply-To: <49B03402.8050601@gmx.net>
References: <49AC6EE4.9080509@gmx.net>
	<8CB69B0FF569DDF-37C-E8A@WEBMAIL-MB10.sysops.aol.com>
	<49AD0F7B.7000802@gmx.net>
	<57E601CD-E7E4-4FA2-B2E4-63B0CED2A877@freeswitch.org>
	<49ADD3E4.20408@gmx.net>
	<8BA7F913-2F0F-4A68-A03A-F4A8973218B9@freeswitch.org>
	<49B03402.8050601@gmx.net>
Message-ID: <87f2f3b90903051232l22db660blba57c46208f42d63@mail.gmail.com>

On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello Brian,
>
> concerning
>> Well you should use ESL then ;)
> I simply do not understand what you mean by this. Is it sarcastic? Am I
> asking stupid questions?
>

ESL = Event Socket Library. It is an abstraction layer to make
interacting with the FS event socket a little easier. Look in the
source directory under libs/esl and you'll see all sorts of stuff.
Also check out the new-but-growing ESL wiki page:

http://wiki.freeswitch.org/wiki/Esl

-MC


From kristian.kielhofner at gmail.com  Thu Mar  5 12:39:27 2009
From: kristian.kielhofner at gmail.com (Kristian Kielhofner)
Date: Thu, 5 Mar 2009 15:39:27 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <131e01c99db9$ed440ba0$c7cc22e0$@net>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
	<49AFC537.45B7.0079.0@averyschools.net>
	<600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
	<131e01c99db9$ed440ba0$c7cc22e0$@net>
Message-ID: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>

A bunch of telephony geeks and a 1900 number - what could go wrong?

Anyways, I too don't understand why people prefer forums.

I follow dozens of mailling lists and a half a dozen e-mail addresses
without ever leaving my mail client.  My mail client happens to be
gmail, btw:

- Much more customization, filtering, etc possible than any "web forum"
- "Local" copies of all messages
- Search is awesome, ever hear of Google? ;)

  Web forums are good when you have to serve ads to people to get
paid.  Other than that they are certainly not the ideal tool for the
job.

  Besides (and don't take this as an insult) - have you ever compared
the web forums to the mailing lists for projects that offer both?  Say
what you want to say about mailing lists and IRC but the reality
(usually) is the l33tz all hang out here and web forums (almost
always) end up with the same groups of n00bz circling around and
around trying to figure out how to accomplish even the most basic of
tasks.

  Obviously that can go both ways but as a rule of thumb the people
that are usually in a position to help others typically prefer mailing
lists (probably for some of the reasons I cited above).  Or maybe they
are just old gray hairs too stuck in their ways.  I don't know. ;)

On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein <damin at nacs.net> wrote:
> You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :)
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com


From mike at jerris.com  Thu Mar  5 13:57:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 5 Mar 2009 16:57:30 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com>
	<49AFBFA6.45B7.0079.0@averyschools.net>
	<CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org>
	<CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca>
	<49AFC537.45B7.0079.0@averyschools.net>
	<600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org>
	<131e01c99db9$ed440ba0$c7cc22e0$@net>
	<2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
Message-ID: <5C6F6B6E-EE30-44CD-996D-ED0A2FD0F894@jerris.com>


On Mar 5, 2009, at 3:39 PM, Kristian Kielhofner wrote:

> A bunch of telephony geeks and a 1900 number - what could go wrong?
>
> Anyways, I too don't understand why people prefer forums.
>
> I follow dozens of mailling lists and a half a dozen e-mail addresses
> without ever leaving my mail client.  My mail client happens to be
> gmail, btw:
>
> - Much more customization, filtering, etc possible than any "web  
> forum"
> - "Local" copies of all messages
> - Search is awesome, ever hear of Google? ;)
>
>  Web forums are good when you have to serve ads to people to get
> paid.  Other than that they are certainly not the ideal tool for the
> job.
>
>  Besides (and don't take this as an insult) - have you ever compared
> the web forums to the mailing lists for projects that offer both?  Say
> what you want to say about mailing lists and IRC but the reality
> (usually) is the l33tz all hang out here and web forums (almost
> always) end up with the same groups of n00bz circling around and
> around trying to figure out how to accomplish even the most basic of
> tasks.
>
>  Obviously that can go both ways but as a rule of thumb the people
> that are usually in a position to help others typically prefer mailing
> lists (probably for some of the reasons I cited above).  Or maybe they
> are just old gray hairs too stuck in their ways.  I don't know. ;)


I previously thought the same about the insiders always preferring  
mailing lists but I have a friend who is a core member of the cacti  
developers and they apparently prefer the forums to the lists, but  
have both.  Food for thought at least.

MIke



From dynaguy at gmail.com  Thu Mar  5 12:19:49 2009
From: dynaguy at gmail.com (Dyna Guy)
Date: Thu, 5 Mar 2009 12:19:49 -0800
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
Message-ID: <e5d501740903051219m2dba7b99v89fa8193c3d9dd2c@mail.gmail.com>

I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh
install) and followed the instruction on the wiki installed the new
Freeswitch v1.0.3.  After the installation I can start FS by issus command
"/usr/local/freeswitch/bin/freeswitch".

After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and
chmod it to 755. Reboot the server and FS is not running.

If I try the start up script:
[root at localhost build]# /etc/init.d/freeswitch start
Starting freeswitch:                                       [  OK  ]
[root at localhost build]# /etc/init.d/freeswitch status
freeswitch dead but subsys locked
Did I miss somthing here? Please help. Thanks a lot.

dynaguy





Here is the copy of the origenal freeswitch.init.centos5
----------------------------

[root at localhost ~]# cat /etc/init.d/freeswitch
#!/bin/bash
#
#       /etc/rc.d/init.d/freeswitch
#
#       The FreeSwitch Open Source Voice Platform
#
#  chkconfig: 345 89 14
#  description: Starts and stops the freeswitch server daemon
#  processname: freeswitch
#  config: /usr/local/freeswitch/conf/freeswitch.conf
#  pidfile: /usr/local/freeswitch/log/freeswitch.pid
#
# Source function library.
. /etc/init.d/functions
PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/usr/local/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS="-nc"
RETVAL=0
# Source options file
if [ -f /etc/sysconfig/freeswitch ]; then
        . /etc/sysconfig/freeswitch
fi
# <define any local shell functions used by the code that follows>
start() {
        echo -n "Starting $PROG_NAME: "
        if [ -e $LOCK_FILE ]; then
            if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then
                echo
                echo -n $"$PROG_NAME is already running.";
                failure $"$PROG_NAME is already running.";
                echo
                return 1
            fi
        fi
        cd $FS_HOME
        daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE
$FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1"
                echo
                RETVAL=$?
        [ $RETVAL -eq 0 ] && touch $LOCK_FILE;
        echo
        return $RETVAL
}
stop() {
        echo -n "Shutting down $PROG_NAME: "
        if [ ! -e $LOCK_FILE ]; then
            echo
            echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running."
            failure $"cannot stop $PROG_NAME: $PROG_NAME is not running."
            echo
            return 1;
        fi
        cd $FS_HOME
        $FS_FILE -stop > /dev/null 2>&1
        killproc $PROG_NAME
        RETVAL=$?
        echo
        [ $RETVAL -eq 0 ] &&  rm -f $LOCK_FILE;
        return $RETVAL
}
rhstatus() {
        status $PROG_NAME;
}
case "$1" in
    start)
        start
        ;;
    stop)
        stop
        ;;
    status)
        status $PROG_NAME
        RETVAL=$?
        ;;
    restart)
        stop
        start
        ;;
    reload)
#        <cause the service configuration to be reread, either with
#        kill -HUP or by restarting the daemons, in a manner similar
#        to restart above>
        ;;
    condrestart)
        [ -f $PID_FILE ] && restart || :
        ;;
    *)
        echo "Usage: $PROG_NAME {start|stop|status|reload|restart}"
        exit 1
        ;;
esac
exit $RETVAL
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From dynaguy at gmail.com  Thu Mar  5 13:09:16 2009
From: dynaguy at gmail.com (dynaguy)
Date: Thu, 5 Mar 2009 13:09:16 -0800
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
Message-ID: <2D9EE07B35234097977448BE8EDCEDA9@dell200>

Hello,

I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh
install) and followed the instruction on the wiki installed the new
Freeswitch v1.0.3.  After the installation I can start FS by issus command
"/usr/local/freeswitch/bin/freeswitch".

 

After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and
chmod it to 755. Reboot the server and FS is not running. 

 

If I try the start up script:

[root at localhost build]# /etc/init.d/freeswitch start
Starting freeswitch:                                       [  OK  ]

[root at localhost build]# /etc/init.d/freeswitch status
freeswitch dead but subsys locked

Did I miss somthing here? Please help. Thanks a lot.

 

dynaguy

 

 

 

 

 

Here is the copy of the origenal freeswitch.init.centos5

----------------------------


[root at localhost ~]# cat /etc/init.d/freeswitch
#!/bin/bash
#
#       /etc/rc.d/init.d/freeswitch
#
#       The FreeSwitch Open Source Voice Platform
#
#  chkconfig: 345 89 14
#  description: Starts and stops the freeswitch server daemon
#  processname: freeswitch
#  config: /usr/local/freeswitch/conf/freeswitch.conf
#  pidfile: /usr/local/freeswitch/log/freeswitch.pid
#

# Source function library.
. /etc/init.d/functions

PROG_NAME=freeswitch
PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
FS_USER=${FS_USER-freeswitch}
FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
FS_HOME=${FS_HOME-/usr/local/freeswitch}
LOCK_FILE=/var/lock/subsys/freeswitch
FREESWITCH_ARGS="-nc"
RETVAL=0

# Source options file
if [ -f /etc/sysconfig/freeswitch ]; then
        . /etc/sysconfig/freeswitch
fi

# <define any local shell functions used by the code that follows>

start() {
        echo -n "Starting $PROG_NAME: "
        if [ -e $LOCK_FILE ]; then
            if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then
                echo
                echo -n $"$PROG_NAME is already running.";
                failure $"$PROG_NAME is already running.";
                echo
                return 1
            fi
        fi
        cd $FS_HOME
        daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE
$FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1"
                echo
                RETVAL=$?
        [ $RETVAL -eq 0 ] && touch $LOCK_FILE;
        echo
        return $RETVAL
}

stop() {
        echo -n "Shutting down $PROG_NAME: "
        if [ ! -e $LOCK_FILE ]; then
            echo
            echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running."
            failure $"cannot stop $PROG_NAME: $PROG_NAME is not running."
            echo
            return 1;
        fi
        cd $FS_HOME
        $FS_FILE -stop > /dev/null 2>&1
        killproc $PROG_NAME
        RETVAL=$?
        echo
        [ $RETVAL -eq 0 ] &&  rm -f $LOCK_FILE;
        return $RETVAL
}

rhstatus() {
        status $PROG_NAME;
}

case "$1" in
    start)
        start
        ;;
    stop)
        stop
        ;;
    status)
        status $PROG_NAME
        RETVAL=$?
        ;;
    restart)
        stop
        start
        ;;
    reload)
#        <cause the service configuration to be reread, either with
#        kill -HUP or by restarting the daemons, in a manner similar
#        to restart above>
        ;;
    condrestart)
        [ -f $PID_FILE ] && restart || :
        ;;
    *)
        echo "Usage: $PROG_NAME {start|stop|status|reload|restart}"
        exit 1
        ;;
esac
exit $RETVAL

 

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From stevecrozz at gmail.com  Thu Mar  5 14:26:27 2009
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Thu, 5 Mar 2009 14:26:27 -0800
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
In-Reply-To: <e5d501740903051219m2dba7b99v89fa8193c3d9dd2c@mail.gmail.com>
References: <e5d501740903051219m2dba7b99v89fa8193c3d9dd2c@mail.gmail.com>
Message-ID: <11990ade0903051426h4390602fy6f92b9020618fba8@mail.gmail.com>

Your script at /etc/init.d/freeswitch is probably not referenced
anywhere in your init sequence. You should read some documentation on
the boot process for your system, which is probably something like
this: http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/ref-guide/s1-boot-init-shutdown-sysv.html

Someone else might have more specifics for you.

--Stephen

On Thu, Mar 5, 2009 at 12:19 PM, Dyna Guy <dynaguy at gmail.com> wrote:
> I am a newbie to FS and I want learn it. So I setup a Centos 5.2 (fresh
> install) and followed the instruction on the wiki installed the new
> Freeswitch v1.0.3.? After the installation I can start FS by issus command
> "/usr/local/freeswitch/bin/freeswitch".
>
> After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch and
> chmod it to 755. Reboot the server and FS is not running.
>
> If I try the start up script:
> [root at localhost build]# /etc/init.d/freeswitch start
> Starting freeswitch:?????????????????????????????????????? [? OK? ]
> [root at localhost build]# /etc/init.d/freeswitch status
> freeswitch dead but subsys locked
> Did I miss somthing here? Please help. Thanks a lot.
>
> dynaguy
>
>
>
>
>
> Here is the copy of the origenal freeswitch.init.centos5
> ----------------------------
> [root at localhost ~]# cat /etc/init.d/freeswitch
> #!/bin/bash
> #
> #?????? /etc/rc.d/init.d/freeswitch
> #
> #?????? The FreeSwitch Open Source Voice Platform
> #
> #? chkconfig: 345 89 14
> #? description: Starts and stops the freeswitch server daemon
> #? processname: freeswitch
> #? config: /usr/local/freeswitch/conf/freeswitch.conf
> #? pidfile: /usr/local/freeswitch/log/freeswitch.pid
> #
> # Source function library.
> . /etc/init.d/functions
> PROG_NAME=freeswitch
> PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
> FS_USER=${FS_USER-freeswitch}
> FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
> FS_HOME=${FS_HOME-/usr/local/freeswitch}
> LOCK_FILE=/var/lock/subsys/freeswitch
> FREESWITCH_ARGS="-nc"
> RETVAL=0
> # Source options file
> if [ -f /etc/sysconfig/freeswitch ]; then
> ??????? . /etc/sysconfig/freeswitch
> fi
> # <define any local shell functions used by the code that follows>
> start() {
> ??????? echo -n "Starting $PROG_NAME: "
> ??????? if [ -e $LOCK_FILE ]; then
> ??????????? if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then
> ??????????????? echo
> ??????????????? echo -n $"$PROG_NAME is already running.";
> ??????????????? failure $"$PROG_NAME is already running.";
> ??????????????? echo
> ??????????????? return 1
> ??????????? fi
> ??????? fi
> ??????? cd $FS_HOME
> ??????? daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE
> $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1"
> ??????????????? echo
> ??????????????? RETVAL=$?
> ??????? [ $RETVAL -eq 0 ] && touch $LOCK_FILE;
> ??????? echo
> ??????? return $RETVAL
> }
> stop() {
> ??????? echo -n "Shutting down $PROG_NAME: "
> ??????? if [ ! -e $LOCK_FILE ]; then
> ??????????? echo
> ??????????? echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not running."
> ??????????? failure $"cannot stop $PROG_NAME: $PROG_NAME is not running."
> ??????????? echo
> ??????????? return 1;
> ??????? fi
> ??????? cd $FS_HOME
> ??????? $FS_FILE -stop > /dev/null 2>&1
> ??????? killproc $PROG_NAME
> ??????? RETVAL=$?
> ??????? echo
> ??????? [ $RETVAL -eq 0 ] &&? rm -f $LOCK_FILE;
> ??????? return $RETVAL
> }
> rhstatus() {
> ??????? status $PROG_NAME;
> }
> case "$1" in
> ??? start)
> ??????? start
> ??????? ;;
> ??? stop)
> ??????? stop
> ??????? ;;
> ??? status)
> ??????? status $PROG_NAME
> ??????? RETVAL=$?
> ??????? ;;
> ??? restart)
> ??????? stop
> ??????? start
> ??????? ;;
> ??? reload)
> #??????? <cause the service configuration to be reread, either with
> #??????? kill -HUP or by restarting the daemons, in a manner similar
> #??????? to restart above>
> ??????? ;;
> ??? condrestart)
> ??????? [ -f $PID_FILE ] && restart || :
> ??????? ;;
> ??? *)
> ??????? echo "Usage: $PROG_NAME {start|stop|status|reload|restart}"
> ??????? exit 1
> ??????? ;;
> esac
> exit $RETVAL
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 e at musinghalfwit.org  Thu Mar  5 14:38:42 2009
From: e at musinghalfwit.org (Eric Liedtke)
Date: Thu, 5 Mar 2009 16:38:42 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
Message-ID: <20090305223842.GA31993@pointone.com>

Greetings, 

I've been using FS in production on this rev (I realize it's pretty far
behind current) and it's been running well, save 1 issue.

The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I have
2 sip profiles created , 1 per ip interface. This is being used to
terminate traffic to a provider so calls are only 1 direction. They come
into the private side profile, get routed via dialplan to the gateway
defined in the external profile and on to the vendor. Pretty simple.

I have noticed that under load (50 or so cps with ~800-900 bridged calls up)
that over time some channels on the public side seem to get "stuck".  Due to
the nature of how this is being used , I would expect both sip profiles to show
the same number of channels in use any time i do a 'sofia status' ( or at least
be within a channel or 2 of each other). However after a day of heavy use I had
a disparity of ~250 channels. These extra channels also seem to put some
continual load on the 'system cpu' as well , reported via top.

Of course due to the load on the box I have to keep logging turned way
down. So I've been trying to troubleshoot it as best I can.

Last night I grabbed a core file and started in with GDB today. I found
the 120 or so threads that represented real active calls when I took the
corefile, I also found ~250 threads that appeared to be stuck in the
CS_NEW state. The backtraces on all of them looks the same, annotated below.

I walked through the code path by hand , based on the bt's and I don't see how 
this could be happening  unless it's a locking issue. But as far as I can tell
each  session  has it's own mutex defined in the switch_core_session_t struct, 
so I wouldn't think they would be stepping on each other. I also would have expected
if it were something of a deadlock nature it would stop processing calls all 
together.

I grabbed the commands from the .gdbinit (super handy btw!!) and have been trolling 
through the variables to try to ascertain something about why these threads seem to 
be stuck, but am not having much luck even coming up with a scenario to try 
to replicate the issue.

If anyone has any pointers as to where I might look next it would be greatly
appreciated. 

We will be updating to the newest release soon, however I was hoping to nail down
what is going so I can systematically replicate it and verify by testing in the lab
that it is fixed , rather than just pushing the new release to produvction and hoping.

Thanks in advance for any tips/pointers anyone may have.

-e

......bt and bt full for a single "hung" thread


#0  0xb7fd5410 in __kernel_vsyscall ()
#1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
#2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
#3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
#4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462
#5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853
#6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138
#7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
(gdb) bt full
#0  0xb7fd5410 in __kernel_vsyscall ()
No symbol table info available.
#1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
No symbol table info available.
#2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
No symbol table info available.
#3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
No locals.
#4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/switch_core_state_machine.c:462
        exception = 0 '\0'
        state = <value optimized out>
        endstate = CS_NEW
        endpoint_interface = <value optimized out>
        driver_state_handler = (const switch_state_handler_table_t *) 0xb73b1720
        application_state_handler = <value optimized out>
        thread_id = 3085554955
        env = {{__jmpbuf = {134603552, -1428248680, -1461722504, 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895, __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280, 9184, 1, 2976641592, 2833244792, 3086590960, 
        168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960, 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728, 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728, 3085458203, 3086590960, 2976606624, 
        134564192, 2833244904}}}}
        sig = <value optimized out>
        __func__ = "switch_core_session_run"
        __PRETTY_FUNCTION__ = "switch_core_session_run"
#5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840, obj=0x95fe270) at src/switch_core_session.c:853
        session = (switch_core_session_t *) 0x95fe270
        event = <value optimized out>
        event_str = 0x0
        val = <value optimized out>
        __func__ = "switch_core_session_thread"
        __PRETTY_FUNCTION__ = "switch_core_session_thread"
#6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/thread.c:138
No locals.
#7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
No symbol table info available.
#8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6



From brian at freeswitch.org  Thu Mar  5 14:52:43 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 16:52:43 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <20090305223842.GA31993@pointone.com>
References: <20090305223842.GA31993@pointone.com>
Message-ID: <CB697105-D6EB-4255-AC01-2DA7A77AE0BE@freeswitch.org>

Well the rules usually state that you try SVN trunk then report a jira  
if the problem persists but since you're 2000+ revs behind chances are  
we already fixed this issue.  Are you using bypass media?

/b

On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote:

> Greetings,
>
> I've been using FS in production on this rev (I realize it's pretty  
> far
> behind current) and it's been running well, save 1 issue.



From mrene_lists at avgs.ca  Thu Mar  5 14:55:33 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 5 Mar 2009 17:55:33 -0500
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <20090305223842.GA31993@pointone.com>
References: <20090305223842.GA31993@pointone.com>
Message-ID: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>

HI,

If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs 
.
This gives the whole team a way of following up on issues.

Also can you upgrade to svn trunk? A lot of fixes gets committed  
daily, so its good to stay up to date.

As you seem familiar with GDB, you may symlink the .gdbinit file in  
the support-d/ folder to your home directory.
This will give you some FS-specific macros such as "list_sessions"  
which will dump a list of uuids to session pointers.

In your jira, make sure you include "thread apply all bt",  
"list_sessions" and show channels (this one goes in FS) but PLEASE  
update to svn trunk and test again to see if it still happens.

Also, are you using proxy/bypass media or just the default?

Math

On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote:

> Greetings,
>
> I've been using FS in production on this rev (I realize it's pretty  
> far
> behind current) and it's been running well, save 1 issue.
>
> The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I  
> have
> 2 sip profiles created , 1 per ip interface. This is being used to
> terminate traffic to a provider so calls are only 1 direction. They  
> come
> into the private side profile, get routed via dialplan to the gateway
> defined in the external profile and on to the vendor. Pretty simple.
>
> I have noticed that under load (50 or so cps with ~800-900 bridged  
> calls up)
> that over time some channels on the public side seem to get  
> "stuck".  Due to
> the nature of how this is being used , I would expect both sip  
> profiles to show
> the same number of channels in use any time i do a 'sofia  
> status' ( or at least
> be within a channel or 2 of each other). However after a day of  
> heavy use I had
> a disparity of ~250 channels. These extra channels also seem to put  
> some
> continual load on the 'system cpu' as well , reported via top.
>
> Of course due to the load on the box I have to keep logging turned way
> down. So I've been trying to troubleshoot it as best I can.
>
> Last night I grabbed a core file and started in with GDB today. I  
> found
> the 120 or so threads that represented real active calls when I took  
> the
> corefile, I also found ~250 threads that appeared to be stuck in the
> CS_NEW state. The backtraces on all of them looks the same,  
> annotated below.
>
> I walked through the code path by hand , based on the bt's and I  
> don't see how
> this could be happening  unless it's a locking issue. But as far as  
> I can tell
> each  session  has it's own mutex defined in the  
> switch_core_session_t struct,
> so I wouldn't think they would be stepping on each other. I also  
> would have expected
> if it were something of a deadlock nature it would stop processing  
> calls all
> together.
>
> I grabbed the commands from the .gdbinit (super handy btw!!) and  
> have been trolling
> through the variables to try to ascertain something about why these  
> threads seem to
> be stuck, but am not having much luck even coming up with a scenario  
> to try
> to replicate the issue.
>
> If anyone has any pointers as to where I might look next it would be  
> greatly
> appreciated.
>
> We will be updating to the newest release soon, however I was hoping  
> to nail down
> what is going so I can systematically replicate it and verify by  
> testing in the lab
> that it is fixed , rather than just pushing the new release to  
> produvction and hoping.
>
> Thanks in advance for any tips/pointers anyone may have.
>
> -e
>
> ......bt and bt full for a single "hung" thread
>
>
> #0  0xb7fd5410 in __kernel_vsyscall ()
> #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ 
> switch_core_state_machine.c:462
> #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> obj=0x95fe270) at src/switch_core_session.c:853
> #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ 
> thread.c:138
> #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> libpthread.so.0
> #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> (gdb) bt full
> #0  0xb7fd5410 in __kernel_vsyscall ()
> No symbol table info available.
> #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> No symbol table info available.
> #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> No symbol table info available.
> #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> No locals.
> #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ 
> switch_core_state_machine.c:462
>        exception = 0 '\0'
>        state = <value optimized out>
>        endstate = CS_NEW
>        endpoint_interface = <value optimized out>
>        driver_state_handler = (const switch_state_handler_table_t *)  
> 0xb73b1720
>        application_state_handler = <value optimized out>
>        thread_id = 3085554955
>        env = {{__jmpbuf = {134603552, -1428248680, -1461722504,  
> 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895,  
> __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280,  
> 9184, 1, 2976641592, 2833244792, 3086590960,
>        168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960,  
> 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728,  
> 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728,  
> 3085458203, 3086590960, 2976606624,
>        134564192, 2833244904}}}}
>        sig = <value optimized out>
>        __func__ = "switch_core_session_run"
>        __PRETTY_FUNCTION__ = "switch_core_session_run"
> #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> obj=0x95fe270) at src/switch_core_session.c:853
>        session = (switch_core_session_t *) 0x95fe270
>        event = <value optimized out>
>        event_str = 0x0
>        val = <value optimized out>
>        __func__ = "switch_core_session_thread"
>        __PRETTY_FUNCTION__ = "switch_core_session_thread"
> #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ 
> thread.c:138
> No locals.
> #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> libpthread.so.0
> No symbol table info available.
> #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 e at musinghalfwit.org  Thu Mar  5 15:19:00 2009
From: e at musinghalfwit.org (Eric Liedtke)
Date: Thu, 5 Mar 2009 17:19:00 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <CB697105-D6EB-4255-AC01-2DA7A77AE0BE@freeswitch.org>
References: <20090305223842.GA31993@pointone.com>
	<CB697105-D6EB-4255-AC01-2DA7A77AE0BE@freeswitch.org>
Message-ID: <20090305231900.GB31993@pointone.com>

Yeah I know ;) I didn't open a bug because my rev was so far behind. I
was just looking for any advice for where to poke next. Troubleshooting
this has been a fantastic introduction to some of the inner workings of
freeswitch so I was hoping to see it through and learn as I went.

To answer your question no we are not using bypass media. 

-e

It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 , Brian West said:
> Well the rules usually state that you try SVN trunk then report a jira  
> if the problem persists but since you're 2000+ revs behind chances are  
> we already fixed this issue.  Are you using bypass media?
> 
> /b
> 
> On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote:
> 
> > Greetings,
> >
> > I've been using FS in production on this rev (I realize it's pretty  
> > far
> > behind current) and it's been running well, save 1 issue.
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 e at musinghalfwit.org  Thu Mar  5 15:22:37 2009
From: e at musinghalfwit.org (Eric Liedtke)
Date: Thu, 5 Mar 2009 17:22:37 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>
References: <20090305223842.GA31993@pointone.com>
	<71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>
Message-ID: <20090305232237.GC31993@pointone.com>

Yup, as I mentioned to brian didn't want to clog jira with a bug that's
been fixed or report against a rev 2k+ revs behind. I was trying to work
through it as a learning exercise. And yeah I actually added a bunch of
stuff to the list_sessions function to spit out a variety of associated
variables for each session looking for a pattern somewhere to clue me
into what might be happening.

No proxy or bypass media here, just defaults.

I will keep at it and once we update the production systems, if the
problem persists I will open a bug in jira with all the neccessary
goodies. 

Thanks
-e

It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM -0500 , Mathieu Rene said:
> HI,
> 
> If you suspect a bug, the place to report it is JIRA. See: http://wiki.freeswitch.org/wiki/Reporting_Bugs 
> .
> This gives the whole team a way of following up on issues.
> 
> Also can you upgrade to svn trunk? A lot of fixes gets committed  
> daily, so its good to stay up to date.
> 
> As you seem familiar with GDB, you may symlink the .gdbinit file in  
> the support-d/ folder to your home directory.
> This will give you some FS-specific macros such as "list_sessions"  
> which will dump a list of uuids to session pointers.
> 
> In your jira, make sure you include "thread apply all bt",  
> "list_sessions" and show channels (this one goes in FS) but PLEASE  
> update to svn trunk and test again to see if it still happens.
> 
> Also, are you using proxy/bypass media or just the default?
> 
> Math
> 
> On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote:
> 
> > Greetings,
> >
> > I've been using FS in production on this rev (I realize it's pretty  
> > far
> > behind current) and it's been running well, save 1 issue.
> >
> > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I  
> > have
> > 2 sip profiles created , 1 per ip interface. This is being used to
> > terminate traffic to a provider so calls are only 1 direction. They  
> > come
> > into the private side profile, get routed via dialplan to the gateway
> > defined in the external profile and on to the vendor. Pretty simple.
> >
> > I have noticed that under load (50 or so cps with ~800-900 bridged  
> > calls up)
> > that over time some channels on the public side seem to get  
> > "stuck".  Due to
> > the nature of how this is being used , I would expect both sip  
> > profiles to show
> > the same number of channels in use any time i do a 'sofia  
> > status' ( or at least
> > be within a channel or 2 of each other). However after a day of  
> > heavy use I had
> > a disparity of ~250 channels. These extra channels also seem to put  
> > some
> > continual load on the 'system cpu' as well , reported via top.
> >
> > Of course due to the load on the box I have to keep logging turned way
> > down. So I've been trying to troubleshoot it as best I can.
> >
> > Last night I grabbed a core file and started in with GDB today. I  
> > found
> > the 120 or so threads that represented real active calls when I took  
> > the
> > corefile, I also found ~250 threads that appeared to be stuck in the
> > CS_NEW state. The backtraces on all of them looks the same,  
> > annotated below.
> >
> > I walked through the code path by hand , based on the bt's and I  
> > don't see how
> > this could be happening  unless it's a locking issue. But as far as  
> > I can tell
> > each  session  has it's own mutex defined in the  
> > switch_core_session_t struct,
> > so I wouldn't think they would be stepping on each other. I also  
> > would have expected
> > if it were something of a deadlock nature it would stop processing  
> > calls all
> > together.
> >
> > I grabbed the commands from the .gdbinit (super handy btw!!) and  
> > have been trolling
> > through the variables to try to ascertain something about why these  
> > threads seem to
> > be stuck, but am not having much luck even coming up with a scenario  
> > to try
> > to replicate the issue.
> >
> > If anyone has any pointers as to where I might look next it would be  
> > greatly
> > appreciated.
> >
> > We will be updating to the newest release soon, however I was hoping  
> > to nail down
> > what is going so I can systematically replicate it and verify by  
> > testing in the lab
> > that it is fixed , rather than just pushing the new release to  
> > produvction and hoping.
> >
> > Thanks in advance for any tips/pointers anyone may have.
> >
> > -e
> >
> > ......bt and bt full for a single "hung" thread
> >
> >
> > #0  0xb7fd5410 in __kernel_vsyscall ()
> > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ 
> > switch_core_state_machine.c:462
> > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> > obj=0x95fe270) at src/switch_core_session.c:853
> > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ 
> > thread.c:138
> > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> > libpthread.so.0
> > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> > (gdb) bt full
> > #0  0xb7fd5410 in __kernel_vsyscall ()
> > No symbol table info available.
> > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > No symbol table info available.
> > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > No symbol table info available.
> > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > No locals.
> > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at src/ 
> > switch_core_state_machine.c:462
> >        exception = 0 '\0'
> >        state = <value optimized out>
> >        endstate = CS_NEW
> >        endpoint_interface = <value optimized out>
> >        driver_state_handler = (const switch_state_handler_table_t *)  
> > 0xb73b1720
> >        application_state_handler = <value optimized out>
> >        thread_id = 3085554955
> >        env = {{__jmpbuf = {134603552, -1428248680, -1461722504,  
> > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895,  
> > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280,  
> > 9184, 1, 2976641592, 2833244792, 3086590960,
> >        168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960,  
> > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728,  
> > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728,  
> > 3085458203, 3086590960, 2976606624,
> >        134564192, 2833244904}}}}
> >        sig = <value optimized out>
> >        __func__ = "switch_core_session_run"
> >        __PRETTY_FUNCTION__ = "switch_core_session_run"
> > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> > obj=0x95fe270) at src/switch_core_session.c:853
> >        session = (switch_core_session_t *) 0x95fe270
> >        event = <value optimized out>
> >        event_str = 0x0
> >        val = <value optimized out>
> >        __func__ = "switch_core_session_thread"
> >        __PRETTY_FUNCTION__ = "switch_core_session_thread"
> > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at threadproc/unix/ 
> > thread.c:138
> > No locals.
> > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> > libpthread.so.0
> > No symbol table info available.
> > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 raul at etellicom.com  Thu Mar  5 15:32:52 2009
From: raul at etellicom.com (Raul Fragoso)
Date: Thu, 05 Mar 2009 20:32:52 -0300
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
In-Reply-To: <2D9EE07B35234097977448BE8EDCEDA9@dell200>
References: <2D9EE07B35234097977448BE8EDCEDA9@dell200>
Message-ID: <1236295972.18566.38.camel@raul-laptop>

Hi, and welcome to FreeSWITCH !

You've done everything right, now you only need to tell your system to
run that init script during startup ;-)
As root, do this:
chkconfig --add freeswitch
chkconfig --level 2345 freeswitch on

That's all.

Regards,

Raul

On Thu, 2009-03-05 at 13:09 -0800, dynaguy wrote:
> Hello,
> 
> I am a newbie to FS and I want learn it. So I setup a Centos 5.2
> (fresh install) and followed the instruction on the wiki installed the
> new Freeswitch v1.0.3.  After the installation I can start FS by issus
> command "/usr/local/freeswitch/bin/freeswitch".
> 
>  
> 
> After that I copied freeswitch.init.centos5 to /etc/init.d/freeswitch
> and chmod it to 755. Reboot the server and FS is not running. 
> 
>  
> 
> If I try the start up script:
> 
> [root at localhost build]# /etc/init.d/freeswitch start
> Starting freeswitch:                                       [  OK  ]
> 
> [root at localhost build]# /etc/init.d/freeswitch status
> freeswitch dead but subsys locked
> 
> Did I miss somthing here? Please help. Thanks a lot.
> 
>  
> 
> dynaguy
> 
>  
> 
>  
> 
>  
> 
>  
> 
>  
> 
> Here is the copy of the origenal freeswitch.init.centos5
> 
> ----------------------------
> 
> 
> [root at localhost ~]# cat /etc/init.d/freeswitch
> #!/bin/bash
> #
> #       /etc/rc.d/init.d/freeswitch
> #
> #       The FreeSwitch Open Source Voice Platform
> #
> #  chkconfig: 345 89 14
> #  description: Starts and stops the freeswitch server daemon
> #  processname: freeswitch
> #  config: /usr/local/freeswitch/conf/freeswitch.conf
> #  pidfile: /usr/local/freeswitch/log/freeswitch.pid
> #
> 
> # Source function library.
> . /etc/init.d/functions
> 
> PROG_NAME=freeswitch
> PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid}
> FS_USER=${FS_USER-freeswitch}
> FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch}
> FS_HOME=${FS_HOME-/usr/local/freeswitch}
> LOCK_FILE=/var/lock/subsys/freeswitch
> FREESWITCH_ARGS="-nc"
> RETVAL=0
> 
> # Source options file
> if [ -f /etc/sysconfig/freeswitch ]; then
>         . /etc/sysconfig/freeswitch
> fi
> 
> # <define any local shell functions used by the code that follows>
> 
> start() {
>         echo -n "Starting $PROG_NAME: "
>         if [ -e $LOCK_FILE ]; then
>             if [ -e $PID_FILE ] && [ -e /proc/`cat $PID_FILE` ]; then
>                 echo
>                 echo -n $"$PROG_NAME is already running.";
>                 failure $"$PROG_NAME is already running.";
>                 echo
>                 return 1
>             fi
>         fi
>         cd $FS_HOME
>         daemon --user $FS_USER --pidfile $PID_FILE "$FS_FILE
> $FREESWITCH_ARGS $FREESWITCH_PARAMS >/dev/null 2>&1"
>                 echo
>                 RETVAL=$?
>         [ $RETVAL -eq 0 ] && touch $LOCK_FILE;
>         echo
>         return $RETVAL
> }
> 
> stop() {
>         echo -n "Shutting down $PROG_NAME: "
>         if [ ! -e $LOCK_FILE ]; then
>             echo
>             echo -n $"cannot stop $PROG_NAME: $PROG_NAME is not
> running."
>             failure $"cannot stop $PROG_NAME: $PROG_NAME is not
> running."
>             echo
>             return 1;
>         fi
>         cd $FS_HOME
>         $FS_FILE -stop > /dev/null 2>&1
>         killproc $PROG_NAME
>         RETVAL=$?
>         echo
>         [ $RETVAL -eq 0 ] &&  rm -f $LOCK_FILE;
>         return $RETVAL
> }
> 
> rhstatus() {
>         status $PROG_NAME;
> }
> 
> case "$1" in
>     start)
>         start
>         ;;
>     stop)
>         stop
>         ;;
>     status)
>         status $PROG_NAME
>         RETVAL=$?
>         ;;
>     restart)
>         stop
>         start
>         ;;
>     reload)
> #        <cause the service configuration to be reread, either with
> #        kill -HUP or by restarting the daemons, in a manner similar
> #        to restart above>
>         ;;
>     condrestart)
>         [ -f $PID_FILE ] && restart || :
>         ;;
>     *)
>         echo "Usage: $PROG_NAME {start|stop|status|reload|restart}"
>         exit 1
>         ;;
> esac
> exit $RETVAL
> 
>  
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 brian at freeswitch.org  Thu Mar  5 15:39:05 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 17:39:05 -0600
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
In-Reply-To: <1236295972.18566.38.camel@raul-laptop>
References: <2D9EE07B35234097977448BE8EDCEDA9@dell200>
	<1236295972.18566.38.camel@raul-laptop>
Message-ID: <F7A634E9-49D4-4E61-A200-BBA90C024DAF@freeswitch.org>

You might not wanna start it in level 2... network might not be up yet.

/b

On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote:

> Hi, and welcome to FreeSWITCH !
>
> You've done everything right, now you only need to tell your system to
> run that init script during startup ;-)
> As root, do this:
> chkconfig --add freeswitch
> chkconfig --level 2345 freeswitch on
>
> That's all.
>
> Regards,
>
> Raul



From nik.middleton at noblesolutions.co.uk  Thu Mar  5 15:39:45 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Thu, 5 Mar 2009 23:39:45 -0000
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <20090305232237.GC31993@pointone.com>
References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>
	<20090305232237.GC31993@pointone.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BC71@sun.Noble.local>

Well if it's any consolation, I have a 4 day ish old copy of SVN and I
have around 200 of these hung calls, though after an hour or so they did
seem to clear.

That said, FS made 138,330 call attempts today, not too shabby, and
through out the call quality was as good as the first one.  Not sure how
to debug this one.

Version: FreeSWITCH Version 1.0.trunk (12276)

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric
Liedtke
Sent: 05 March 2009 23:23
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231)

Yup, as I mentioned to brian didn't want to clog jira with a bug that's
been fixed or report against a rev 2k+ revs behind. I was trying to work
through it as a learning exercise. And yeah I actually added a bunch of
stuff to the list_sessions function to spit out a variety of associated
variables for each session looking for a pattern somewhere to clue me
into what might be happening.

No proxy or bypass media here, just defaults.

I will keep at it and once we update the production systems, if the
problem persists I will open a bug in jira with all the neccessary
goodies. 

Thanks
-e

It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM
-0500 , Mathieu Rene said:
> HI,
> 
> If you suspect a bug, the place to report it is JIRA. See:
http://wiki.freeswitch.org/wiki/Reporting_Bugs 
> .
> This gives the whole team a way of following up on issues.
> 
> Also can you upgrade to svn trunk? A lot of fixes gets committed  
> daily, so its good to stay up to date.
> 
> As you seem familiar with GDB, you may symlink the .gdbinit file in  
> the support-d/ folder to your home directory.
> This will give you some FS-specific macros such as "list_sessions"  
> which will dump a list of uuids to session pointers.
> 
> In your jira, make sure you include "thread apply all bt",  
> "list_sessions" and show channels (this one goes in FS) but PLEASE  
> update to svn trunk and test again to see if it still happens.
> 
> Also, are you using proxy/bypass media or just the default?
> 
> Math
> 
> On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote:
> 
> > Greetings,
> >
> > I've been using FS in production on this rev (I realize it's pretty

> > far
> > behind current) and it's been running well, save 1 issue.
> >
> > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I  
> > have
> > 2 sip profiles created , 1 per ip interface. This is being used to
> > terminate traffic to a provider so calls are only 1 direction. They

> > come
> > into the private side profile, get routed via dialplan to the
gateway
> > defined in the external profile and on to the vendor. Pretty simple.
> >
> > I have noticed that under load (50 or so cps with ~800-900 bridged  
> > calls up)
> > that over time some channels on the public side seem to get  
> > "stuck".  Due to
> > the nature of how this is being used , I would expect both sip  
> > profiles to show
> > the same number of channels in use any time i do a 'sofia  
> > status' ( or at least
> > be within a channel or 2 of each other). However after a day of  
> > heavy use I had
> > a disparity of ~250 channels. These extra channels also seem to put

> > some
> > continual load on the 'system cpu' as well , reported via top.
> >
> > Of course due to the load on the box I have to keep logging turned
way
> > down. So I've been trying to troubleshoot it as best I can.
> >
> > Last night I grabbed a core file and started in with GDB today. I  
> > found
> > the 120 or so threads that represented real active calls when I took

> > the
> > corefile, I also found ~250 threads that appeared to be stuck in the
> > CS_NEW state. The backtraces on all of them looks the same,  
> > annotated below.
> >
> > I walked through the code path by hand , based on the bt's and I  
> > don't see how
> > this could be happening  unless it's a locking issue. But as far as

> > I can tell
> > each  session  has it's own mutex defined in the  
> > switch_core_session_t struct,
> > so I wouldn't think they would be stepping on each other. I also  
> > would have expected
> > if it were something of a deadlock nature it would stop processing  
> > calls all
> > together.
> >
> > I grabbed the commands from the .gdbinit (super handy btw!!) and  
> > have been trolling
> > through the variables to try to ascertain something about why these

> > threads seem to
> > be stuck, but am not having much luck even coming up with a scenario

> > to try
> > to replicate the issue.
> >
> > If anyone has any pointers as to where I might look next it would be

> > greatly
> > appreciated.
> >
> > We will be updating to the newest release soon, however I was hoping

> > to nail down
> > what is going so I can systematically replicate it and verify by  
> > testing in the lab
> > that it is fixed , rather than just pushing the new release to  
> > produvction and hoping.
> >
> > Thanks in advance for any tips/pointers anyone may have.
> >
> > -e
> >
> > ......bt and bt full for a single "hung" thread
> >
> >
> > #0  0xb7fd5410 in __kernel_vsyscall ()
> > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at
src/ 
> > switch_core_state_machine.c:462
> > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> > obj=0x95fe270) at src/switch_core_session.c:853
> > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at
threadproc/unix/ 
> > thread.c:138
> > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> > libpthread.so.0
> > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> > (gdb) bt full
> > #0  0xb7fd5410 in __kernel_vsyscall ()
> > No symbol table info available.
> > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > No symbol table info available.
> > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > No symbol table info available.
> > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > No locals.
> > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at
src/ 
> > switch_core_state_machine.c:462
> >        exception = 0 '\0'
> >        state = <value optimized out>
> >        endstate = CS_NEW
> >        endpoint_interface = <value optimized out>
> >        driver_state_handler = (const switch_state_handler_table_t *)

> > 0xb73b1720
> >        application_state_handler = <value optimized out>
> >        thread_id = 3085554955
> >        env = {{__jmpbuf = {134603552, -1428248680, -1461722504,  
> > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895,  
> > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280,  
> > 9184, 1, 2976641592, 2833244792, 3086590960,
> >        168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960,

> > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728,  
> > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728,

> > 3085458203, 3086590960, 2976606624,
> >        134564192, 2833244904}}}}
> >        sig = <value optimized out>
> >        __func__ = "switch_core_session_run"
> >        __PRETTY_FUNCTION__ = "switch_core_session_run"
> > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,  
> > obj=0x95fe270) at src/switch_core_session.c:853
> >        session = (switch_core_session_t *) 0x95fe270
> >        event = <value optimized out>
> >        event_str = 0x0
> >        val = <value optimized out>
> >        __func__ = "switch_core_session_thread"
> >        __PRETTY_FUNCTION__ = "switch_core_session_thread"
> > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at
threadproc/unix/ 
> > thread.c:138
> > No locals.
> > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/ 
> > libpthread.so.0
> > No symbol table info available.
> > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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 brian at freeswitch.org  Thu Mar  5 15:43:35 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Mar 2009 17:43:35 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC71@sun.Noble.local>
References: <20090305223842.GA31993@pointone.com><71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>
	<20090305232237.GC31993@pointone.com>
	<A2B4F48489739E439D8845296A2FD9C408BC71@sun.Noble.local>
Message-ID: <063AB580-5ABD-44B8-9671-25FBE1A0483C@freeswitch.org>

I would update... We fixed a few bugs related to hung calls in the  
past 24 hours.

/b

On Mar 5, 2009, at 5:39 PM, Nik Middleton wrote:

> Well if it's any consolation, I have a 4 day ish old copy of SVN and I
> have around 200 of these hung calls, though after an hour or so they  
> did
> seem to clear.
>
> That said, FS made 138,330 call attempts today, not too shabby, and
> through out the call quality was as good as the first one.  Not sure  
> how
> to debug this one.
>
> Version: FreeSWITCH Version 1.0.trunk (12276)



From nik.middleton at noblesolutions.co.uk  Thu Mar  5 15:59:44 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Thu, 5 Mar 2009 23:59:44 -0000
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>

Just curious here.

 

I've always followed the fedora route but became disillusioned with the
focus on the desktop rather than the server mode.  Of late I've moved my
servers to Centos.  I felt the need for stable systems. 

 

Everyone seems to slate Centos, but to my surprise Anthony recommends
Centos 5.2 which is nice to hear.  Yes I know it's not bleeding edge,
but I don't want that.

 

Any reason why I should not be running Centos with FS? (I do plan on
running 64 bit in future though)

 

Regards,

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From mszlazak at aol.com  Thu Mar  5 16:35:50 2009
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Thu, 05 Mar 2009 19:35:50 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
References: <6ec892d20903050214n2e8a0fabyf1b995b120c5a50e@mail.gmail.com><49AFBFA6.45B7.0079.0@averyschools.net><CEFA9CCD-689E-4445-AE20-30CC2D483F15@freeswitch.org><CCCD9A7C-7CF2-42B7-BC13-301839592A60@avgs.ca><49AFC537.45B7.0079.0@averyschools.net><600D5F4A-865B-4F20-871A-D5214B616AC0@freeswitch.org><131e01c99db9$ed440ba0$c7cc22e0$@net>
	<2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
Message-ID: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>

If as you say "people prefer forums" then that's the nature of the target market and controlling markets can be very difficult. So you go with the market to succeed. The "build it and they will come" attitude virtually never works well.



-----Original Message-----
From: Kristian Kielhofner <kristian.kielhofner at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 5 Mar 2009 12:39 pm
Subject: Re: [Freeswitch-users] Please end the torment








A bunch of telephony geeks and a 1900 number - what could go wrong?

Anyways, I too don't understand why people prefer forums.

I follow dozens of mailling lists and a half a dozen e-mail addresses
without ever leaving my mail client.  My mail client happens to be
gmail, btw:

- Much more customization, filtering, etc possible than any "web forum"
- "Local" copies of all messages
- Search is awesome, ever hear of Google? ;)

  Web forums are good when you have to serve ads to people to get
paid.  Other than that they are certainly not the ideal tool for the
job.

  Besides (and don't take this as an insult) - have you ever compared
the web forums to the mailing lists for projects that offer both?  Say
what you want to say about mailing lists and IRC but the reality
(usually) is the l33tz all hang out here and web forums (almost
always) end up with the same groups of n00bz circling around and
around trying to figure out how to accomplish even the most basic of
tasks.

  Obviously that can go both ways but as a rule of thumb the people
that are usually in a position to help others typically prefer mailing
lists (probably for some of the reasons I cited above).  Or maybe they
are just old gray hairs too stuck in their ways.  I don't know. ;)

On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein <damin at nacs.net> wrote:
> You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :)
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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 Mar  5 16:52:36 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 5 Mar 2009 16:52:36 -0800
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
Message-ID: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>

> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos
> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t
> want that.

Repeat the mantra: CentOS is boring and predictable; boring and
predictable is perfect for real-time telephony systems.

> Any reason why I should not be running Centos with FS? (I do plan on running
> 64 bit in future though)

None that I can think of unless you have a super cool Linux distro
that none of us have ever heard of.

64 bit OS on 64 bit hardware is a good thing. :)
-MC


From jason at jasonjgw.net  Thu Mar  5 17:01:24 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 6 Mar 2009 12:01:24 +1100
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
	<8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
Message-ID: <20090306010124.GA12699@jdc.jasonjgw.net>

mszlazak at aol.com <mszlazak at aol.com> wrote:
> If as you say "people prefer forums" then that's the nature of the target
> market and controlling markets can be very difficult. So you go with the
> market to succeed. 

Most free software/open-source people I've encountered prefer mailing lists
and don't like being forced to use a Web interface instead (unless it's the
Web interface of their preferred Web mail provider, in which case they're not
being compelled to use it).

For some of us, a Web forum is hard and inconvenient to use, because it
substitutes the forum operator's user interface for that of the user's
preferred mail client. I have reasons for choosing the mail client that I use,
and if I had to work via somebody else's Web interface instead it would
probably result in my not participating at all.

This list can also be accessed via the Web and over NNTP at gmane.org.

For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user -
just connect your news reader to news.gmane.org.

You can also post from the newsgroup; the first time you do so, an automated
e-mail message will arrive in your inbox requesting confirmation, for spam
prevention purposes.

I don't know whether it is possible to post from the gmane.org Web site. They
use Xapian as their search tool, which, in my experience, usually places the
most relevant posts near the top of the search results.



From dynaguy at gmail.com  Thu Mar  5 17:07:21 2009
From: dynaguy at gmail.com (Dyna Guy)
Date: Thu, 5 Mar 2009 17:07:21 -0800
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
In-Reply-To: <F7A634E9-49D4-4E61-A200-BBA90C024DAF@freeswitch.org>
References: <2D9EE07B35234097977448BE8EDCEDA9@dell200>
	<1236295972.18566.38.camel@raul-laptop>
	<F7A634E9-49D4-4E61-A200-BBA90C024DAF@freeswitch.org>
Message-ID: <e5d501740903051707g3c248f10p445d6caa247f61d3@mail.gmail.com>

Thanks for all your advices.

I am still struggle to make FS start.

I tried few things included:
chkconfig --add freeswitch
chkconfig --level 345 freeswitch on

I also added a user "freeswitch"

The problem is : if I run "/etc/init.d./freeswitch" manually, it says [OK]
like this:
[root at localhost build]# /etc/init.d/freeswitch start
Starting freeswitch:             [  OK  ]

But then I did a "ps aux | grep freeswitch" it doesn't show FS running.

I am not a script guru. If I run FS from commandline as root:
 /usr/local/freeswitch/bin/freeswitch
then I can see FS running.

What did I missing here?

dynaguy






On Thu, Mar 5, 2009 at 3:39 PM, Brian West <brian at freeswitch.org> wrote:

> You might not wanna start it in level 2... network might not be up yet.
>
> /b
>
> On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote:
>
> > Hi, and welcome to FreeSWITCH !
> >
> > You've done everything right, now you only need to tell your system to
> > run that init script during startup ;-)
> > As root, do this:
> > chkconfig --add freeswitch
> > chkconfig --level 2345 freeswitch on
> >
> > That's all.
> >
> > Regards,
> >
> > Raul
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 raul at etellicom.com  Thu Mar  5 17:11:56 2009
From: raul at etellicom.com (Raul Fragoso)
Date: Thu, 05 Mar 2009 22:11:56 -0300
Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot
In-Reply-To: <F7A634E9-49D4-4E61-A200-BBA90C024DAF@freeswitch.org>
References: <2D9EE07B35234097977448BE8EDCEDA9@dell200>
	<1236295972.18566.38.camel@raul-laptop>
	<F7A634E9-49D4-4E61-A200-BBA90C024DAF@freeswitch.org>
Message-ID: <1236301916.18566.39.camel@raul-laptop>

Ah yes, my mistake, thanks for the correction Brian.

On Thu, 2009-03-05 at 17:39 -0600, Brian West wrote:
> You might not wanna start it in level 2... network might not be up yet.
> 
> /b
> 
> On Mar 5, 2009, at 5:32 PM, Raul Fragoso wrote:
> 
> > Hi, and welcome to FreeSWITCH !
> >
> > You've done everything right, now you only need to tell your system to
> > run that init script during startup ;-)
> > As root, do this:
> > chkconfig --add freeswitch
> > chkconfig --level 2345 freeswitch on
> >
> > That's all.
> >
> > Regards,
> >
> > Raul
> 
> 
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 davidwdan at gmail.com  Thu Mar  5 17:26:03 2009
From: davidwdan at gmail.com (David Dan)
Date: Thu, 5 Mar 2009 20:26:03 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <20090306010124.GA12699@jdc.jasonjgw.net>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
	<8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
	<20090306010124.GA12699@jdc.jasonjgw.net>
Message-ID: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>

Web forums are like the wild west of the internet.  They offer nothing that
a mailing list and good wiki can't handle.

Just go take a look at the trixbox forums. The last thing you want it for
someone that is looking into freeswitch for the first time, to come across
something like this (The Beginning of the End for
CE<http://trixbox.org/forums/trixbox-forums/open-discussion/begining-end-ce>),
1 click off the front page.  I'd really hate to see FS go down this Mob Rule
path.




On Thu, Mar 5, 2009 at 8:01 PM, Jason White <jason at jasonjgw.net> wrote:

> mszlazak at aol.com <mszlazak at aol.com> wrote:
> > If as you say "people prefer forums" then that's the nature of the target
> > market and controlling markets can be very difficult. So you go with the
> > market to succeed.
>
> Most free software/open-source people I've encountered prefer mailing lists
> and don't like being forced to use a Web interface instead (unless it's the
> Web interface of their preferred Web mail provider, in which case they're
> not
> being compelled to use it).
>
> For some of us, a Web forum is hard and inconvenient to use, because it
> substitutes the forum operator's user interface for that of the user's
> preferred mail client. I have reasons for choosing the mail client that I
> use,
> and if I had to work via somebody else's Web interface instead it would
> probably result in my not participating at all.
>
> This list can also be accessed via the Web and over NNTP at gmane.org.
>
> For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user
> -
> just connect your news reader to news.gmane.org.
>
> You can also post from the newsgroup; the first time you do so, an
> automated
> e-mail message will arrive in your inbox requesting confirmation, for spam
> prevention purposes.
>
> I don't know whether it is possible to post from the gmane.org Web site.
> They
> use Xapian as their search tool, which, in my experience, usually places
> the
> most relevant posts near the top of the search results.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Thu Mar  5 18:00:05 2009
From: dujinfang at gmail.com (seven)
Date: Fri, 6 Mar 2009 10:00:05 +0800
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <49AFCD02.2000603@laposte.net>
References: <49AFC1C3.9030603@laposte.net>	<165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>
	<49AFC9BB.9090106@laposte.net> <49AFCD02.2000603@laposte.net>
Message-ID: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com>

try  
bridge 
({effective_caller_id_name 
="your_name",effective_caller_id_number="0000"}sofia/b-leg)

On Mar 5, 2009, at 9:00 PM, rod wrote:

> the A leg invite looks like this:
> From: "Anonymous"<sip:anonymous at anonymous.invalid;user=phone>
>
> it has been rewritten like this:
> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>
>
> rod
>
> rod wrote:
>> Hi Brian,
>>
>> if I use the function effective_caller_id_number with my INVITE, I  
>> get this:
>>
>> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS.
>>
>> this is not exactly what I'm looking for :p
>>
>> rod
>>
>>
>> Brian West wrote:
>>
>>> Well this depends on how you're placing the call.. if its a standard
>>> bridge you can on the A-Leg set
>>> "effective_caller_id_number=000${caller_id_number}" before you call
>>> bridge.
>>>
>>> Is the from already in the correct format?
>>>
>>> /b
>>>
>>> On Mar 5, 2009, at 6:12 AM, rod wrote:
>>>
>>>
>>>> Dear list,
>>>>
>>>> I'd like to rewrite the number in the Remote Party ID header and  
>>>> only in
>>>> this header.
>>>>
>>>> ex: I'd like to prefix the caller  ID with a prefix code (000 in  
>>>> this
>>>> example) in the  RPID header :
>>>>
>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>       Remote-Party-ID:
>>>> <sip: 
>>>> 123456 
>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>
>>>> should become:
>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>       Remote-Party-ID:
>>>> <sip: 
>>>> 000123456 
>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>
>>>> But the From field should remain unchanged.
>>>>
>>>> And how to strip this prefix:
>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>       Remote-Party-ID:
>>>> <sip: 
>>>> 000123456 
>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>
>>>> should become:
>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>       Remote-Party-ID:
>>>> <sip: 
>>>> 123456 
>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>
>>>>
>>>> regards.
>>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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  Thu Mar  5 19:59:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Mar 2009 21:59:34 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC71@sun.Noble.local>
References: <20090305223842.GA31993@pointone.com>
	<71DCDD45-0811-4C6F-ACD7-063726F0639F@avgs.ca>
	<20090305232237.GC31993@pointone.com>
	<A2B4F48489739E439D8845296A2FD9C408BC71@sun.Noble.local>
Message-ID: <191c3a030903051959y4a89aafaw108f41648215b35e@mail.gmail.com>

if they went away by themselves they must not have been hung?

On Thu, Mar 5, 2009 at 5:39 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

> Well if it's any consolation, I have a 4 day ish old copy of SVN and I
> have around 200 of these hung calls, though after an hour or so they did
> seem to clear.
>
> That said, FS made 138,330 call attempts today, not too shabby, and
> through out the call quality was as good as the first one.  Not sure how
> to debug this one.
>
> Version: FreeSWITCH Version 1.0.trunk (12276)
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric
> Liedtke
> Sent: 05 March 2009 23:23
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hung Channels (SVN Rev 10231)
>
> Yup, as I mentioned to brian didn't want to clog jira with a bug that's
> been fixed or report against a rev 2k+ revs behind. I was trying to work
> through it as a learning exercise. And yeah I actually added a bunch of
> stuff to the list_sessions function to spit out a variety of associated
> variables for each session looking for a pattern somewhere to clue me
> into what might be happening.
>
> No proxy or bypass media here, just defaults.
>
> I will keep at it and once we update the production systems, if the
> problem persists I will open a bug in jira with all the neccessary
> goodies.
>
> Thanks
> -e
>
> It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 05:55:33PM
> -0500 , Mathieu Rene said:
> > HI,
> >
> > If you suspect a bug, the place to report it is JIRA. See:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs
> > .
> > This gives the whole team a way of following up on issues.
> >
> > Also can you upgrade to svn trunk? A lot of fixes gets committed
> > daily, so its good to stay up to date.
> >
> > As you seem familiar with GDB, you may symlink the .gdbinit file in
> > the support-d/ folder to your home directory.
> > This will give you some FS-specific macros such as "list_sessions"
> > which will dump a list of uuids to session pointers.
> >
> > In your jira, make sure you include "thread apply all bt",
> > "list_sessions" and show channels (this one goes in FS) but PLEASE
> > update to svn trunk and test again to see if it still happens.
> >
> > Also, are you using proxy/bypass media or just the default?
> >
> > Math
> >
> > On 5-Mar-09, at 5:38 PM, Eric Liedtke wrote:
> >
> > > Greetings,
> > >
> > > I've been using FS in production on this rev (I realize it's pretty
>
> > > far
> > > behind current) and it's been running well, save 1 issue.
> > >
> > > The basic setup is an SBC , 2 GiG-E ports, 1 public , 1 private. I
> > > have
> > > 2 sip profiles created , 1 per ip interface. This is being used to
> > > terminate traffic to a provider so calls are only 1 direction. They
>
> > > come
> > > into the private side profile, get routed via dialplan to the
> gateway
> > > defined in the external profile and on to the vendor. Pretty simple.
> > >
> > > I have noticed that under load (50 or so cps with ~800-900 bridged
> > > calls up)
> > > that over time some channels on the public side seem to get
> > > "stuck".  Due to
> > > the nature of how this is being used , I would expect both sip
> > > profiles to show
> > > the same number of channels in use any time i do a 'sofia
> > > status' ( or at least
> > > be within a channel or 2 of each other). However after a day of
> > > heavy use I had
> > > a disparity of ~250 channels. These extra channels also seem to put
>
> > > some
> > > continual load on the 'system cpu' as well , reported via top.
> > >
> > > Of course due to the load on the box I have to keep logging turned
> way
> > > down. So I've been trying to troubleshoot it as best I can.
> > >
> > > Last night I grabbed a core file and started in with GDB today. I
> > > found
> > > the 120 or so threads that represented real active calls when I took
>
> > > the
> > > corefile, I also found ~250 threads that appeared to be stuck in the
> > > CS_NEW state. The backtraces on all of them looks the same,
> > > annotated below.
> > >
> > > I walked through the code path by hand , based on the bt's and I
> > > don't see how
> > > this could be happening  unless it's a locking issue. But as far as
>
> > > I can tell
> > > each  session  has it's own mutex defined in the
> > > switch_core_session_t struct,
> > > so I wouldn't think they would be stepping on each other. I also
> > > would have expected
> > > if it were something of a deadlock nature it would stop processing
> > > calls all
> > > together.
> > >
> > > I grabbed the commands from the .gdbinit (super handy btw!!) and
> > > have been trolling
> > > through the variables to try to ascertain something about why these
>
> > > threads seem to
> > > be stuck, but am not having much luck even coming up with a scenario
>
> > > to try
> > > to replicate the issue.
> > >
> > > If anyone has any pointers as to where I might look next it would be
>
> > > greatly
> > > appreciated.
> > >
> > > We will be updating to the newest release soon, however I was hoping
>
> > > to nail down
> > > what is going so I can systematically replicate it and verify by
> > > testing in the lab
> > > that it is fixed , rather than just pushing the new release to
> > > produvction and hoping.
> > >
> > > Thanks in advance for any tips/pointers anyone may have.
> > >
> > > -e
> > >
> > > ......bt and bt full for a single "hung" thread
> > >
> > >
> > > #0  0xb7fd5410 in __kernel_vsyscall ()
> > > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at
> src/
> > > switch_core_state_machine.c:462
> > > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,
> > > obj=0x95fe270) at src/switch_core_session.c:853
> > > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at
> threadproc/unix/
> > > thread.c:138
> > > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/
> > > libpthread.so.0
> > > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> > > (gdb) bt full
> > > #0  0xb7fd5410 in __kernel_vsyscall ()
> > > No symbol table info available.
> > > #1  0xb7d14cb6 in nanosleep () from /lib/tls/i686/cmov/libc.so.6
> > > No symbol table info available.
> > > #2  0xb7d4f1dc in usleep () from /lib/tls/i686/cmov/libc.so.6
> > > No symbol table info available.
> > > #3  0xb7ee02cd in switch_sleep (t=1000) at src/switch_time.c:143
> > > No locals.
> > > #4  0xb7e9da03 in switch_core_session_run (session=0x95fe270) at
> src/
> > > switch_core_state_machine.c:462
> > >        exception = 0 '\0'
> > >        state = <value optimized out>
> > >        endstate = CS_NEW
> > >        endpoint_interface = <value optimized out>
> > >        driver_state_handler = (const switch_state_handler_table_t *)
>
> > > 0xb73b1720
> > >        application_state_handler = <value optimized out>
> > >        thread_id = 3085554955
> > >        env = {{__jmpbuf = {134603552, -1428248680, -1461722504,
> > > 9184, -1210273432, -1210014020}, __mask_was_saved = -1210034895,
> > > __saved_mask = {__val = {0, 3084988404, 3084937740, 3086469280,
> > > 9184, 1, 2976641592, 2833244792, 3086590960,
> > >        168036728, 3084937740, 2833244808, 3085923728, 1, 3086590960,
>
> > > 2833244840, 3086590960, 0, 134564192, 2833244840, 3085923728,
> > > 134564244, 3086590960, 2833244872, 3085887870, 134564240, 168036728,
>
> > > 3085458203, 3086590960, 2976606624,
> > >        134564192, 2833244904}}}}
> > >        sig = <value optimized out>
> > >        __func__ = "switch_core_session_run"
> > >        __PRETTY_FUNCTION__ = "switch_core_session_run"
> > > #5  0xb7e9c765 in switch_core_session_thread (thread=0x9ada840,
> > > obj=0x95fe270) at src/switch_core_session.c:853
> > >        session = (switch_core_session_t *) 0x95fe270
> > >        event = <value optimized out>
> > >        event_str = 0x0
> > >        val = <value optimized out>
> > >        __func__ = "switch_core_session_thread"
> > >        __PRETTY_FUNCTION__ = "switch_core_session_thread"
> > > #6  0xb7efd916 in dummy_worker (opaque=0x9ada840) at
> threadproc/unix/
> > > thread.c:138
> > > No locals.
> > > #7  0xb7e034fb in start_thread () from /lib/tls/i686/cmov/
> > > libpthread.so.0
> > > No symbol table info available.
> > > #8  0xb7d55e5e in clone () from /lib/tls/i686/cmov/libc.so.6
> > >
> > >
> > > _______________________________________________
> > > Freeswitch-users mailing list
> > > Freeswitch-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Mar  5 20:02:34 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 5 Mar 2009 22:02:34 -0600
Subject: [Freeswitch-users] Hung Channels (SVN Rev 10231)
In-Reply-To: <20090305231900.GB31993@pointone.com>
References: <20090305223842.GA31993@pointone.com>
	<CB697105-D6EB-4255-AC01-2DA7A77AE0BE@freeswitch.org>
	<20090305231900.GB31993@pointone.com>
Message-ID: <191c3a030903052002s2db4e53bi94e489a376b0ad18@mail.gmail.com>

in your case you will have no choice but to update.
Please do a fresh checkout as the build system has also drastically changed.


On Thu, Mar 5, 2009 at 5:19 PM, Eric Liedtke <e at musinghalfwit.org> wrote:

> Yeah I know ;) I didn't open a bug because my rev was so far behind. I
> was just looking for any advice for where to poke next. Troubleshooting
> this has been a fantastic introduction to some of the inner workings of
> freeswitch so I was hoping to see it through and learn as I went.
>
> To answer your question no we are not using bypass media.
>
> -e
>
> It's seems fuzzy now but I think on Thu, Mar 05, 2009 at 04:52:43PM -0600 ,
> Brian West said:
> > Well the rules usually state that you try SVN trunk then report a jira
> > if the problem persists but since you're 2000+ revs behind chances are
> > we already fixed this issue.  Are you using bypass media?
> >
> > /b
> >
> > On Mar 5, 2009, at 4:38 PM, Eric Liedtke wrote:
> >
> > > Greetings,
> > >
> > > I've been using FS in production on this rev (I realize it's pretty
> > > far
> > > behind current) and it's been running well, save 1 issue.
> >
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mszlazak at aol.com  Thu Mar  5 22:39:50 2009
From: mszlazak at aol.com (mszlazak at aol.com)
Date: Fri, 06 Mar 2009 01:39:50 -0500
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net>
	<65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
Message-ID: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>


 Again, if your target market prefers lists, then go with list. If they prefer forums then it's forums. The point is that it's not about what a few like, it's about the mob but the right mob.


 


 

-----Original Message-----
From: David Dan <davidwdan at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, 5 Mar 2009 5:26 pm
Subject: Re: [Freeswitch-users] Please end the torment









Web forums are like the wild west of the internet.? They offer nothing that a mailing list and good wiki can't handle.

Just
go take a look at the trixbox forums. The last thing you want it for
someone that is looking into freeswitch for the first time, to come
across something like this (The Beginning of the End for CE), 1 click off the front page.? I'd really hate to see FS go down this Mob Rule path.






On Thu, Mar 5, 2009 at 8:01 PM, Jason White <jason at jasonjgw.net> wrote:



mszlazak at aol.com <mszlazak at aol.com> wrote:

> If as you say "people prefer forums" then that's the nature of the target

> market and controlling markets can be very difficult. So you go with the

> market to succeed.




Most free software/open-source people I've encountered prefer mailing lists

and don't like being forced to use a Web interface instead (unless it's the

Web interface of their preferred Web mail provider, in which case they're not

being compelled to use it).



For some of us, a Web forum is hard and inconvenient to use, because it

substitutes the forum operator's user interface for that of the user's

preferred mail client. I have reasons for choosing the mail client that I use,

and if I had to work via somebody else's Web interface instead it would

probably result in my not participating at all.



This list can also be accessed via the Web and over NNTP at gmane.org.



For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user -

just connect your news reader to news.gmane.org.



You can also post from the newsgroup; the first time you do so, an automated

e-mail message will arrive in your inbox requesting confirmation, for spam

prevention purposes.



I don't know whether it is possible to post from the gmane.org Web site. They

use Xapian as their search tool, which, in my experience, usually places the

most relevant posts near the top of the search results.









_______________________________________________

Freeswitch-users mailing list

Freeswitch-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org









 





_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at 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 mashudiflexi at telkom.co.id  Thu Mar  5 23:02:45 2009
From: mashudiflexi at telkom.co.id (mashudi)
Date: Fri, 06 Mar 2009 14:02:45 +0700
Subject: [Freeswitch-users] make freeswitch-snapshot
Message-ID: <49B0CA95.5080101@telkom.co.id>

Hi Folk,
i got error while conduct  ./make  freeswitch-snapshot on debian 2.6 x86
here is the error :

/usr/bin/ld: cannot find -lodbc
collect2: ld returned 1 exit status
make[2]: *** [libfreeswitch.la] Error 1
Making all in src
Making all in mod

making all mod_amr
make[5]: *** No rule to make target 
`/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by 
`mod_amr.so'.  Stop.
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Did I miss something ?
thank you for your support.

mashudi



*****************************************
Sekarang Gratis Nelpon SLJJ Flexi diperluas  ke
Yogya
*****************************************


From jason at jasonjgw.net  Thu Mar  5 23:00:36 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 6 Mar 2009 18:00:36 +1100
Subject: [Freeswitch-users] make freeswitch-snapshot
In-Reply-To: <49B0CA95.5080101@telkom.co.id>
References: <49B0CA95.5080101@telkom.co.id>
Message-ID: <20090306070036.GA24314@jdc.jasonjgw.net>

mashudi <mashudiflexi at telkom.co.id> wrote:
> i got error while conduct  ./make  freeswitch-snapshot on debian 2.6 x86
> here is the error :
> 
> /usr/bin/ld: cannot find -lodbc

Have you installed the ODBC library and its development headers? Are they the
latest version?

It's failing to find the ODBC library.



From stevecrozz at gmail.com  Thu Mar  5 23:03:51 2009
From: stevecrozz at gmail.com (Stephen Crosby)
Date: Thu, 5 Mar 2009 23:03:51 -0800
Subject: [Freeswitch-users] make freeswitch-snapshot
In-Reply-To: <20090306070036.GA24314@jdc.jasonjgw.net>
References: <49B0CA95.5080101@telkom.co.id>
	<20090306070036.GA24314@jdc.jasonjgw.net>
Message-ID: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com>

I think you need to install the debian package 'unixodbc-dev'

--Stephen

On Thu, Mar 5, 2009 at 11:00 PM, Jason White <jason at jasonjgw.net> wrote:
> mashudi <mashudiflexi at telkom.co.id> wrote:
>> i got error while conduct ?./make ?freeswitch-snapshot on debian 2.6 x86
>> here is the error :
>>
>> /usr/bin/ld: cannot find -lodbc
>
> Have you installed the ODBC library and its development headers? Are they the
> latest version?
>
> It's failing to find the ODBC library.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 kawarod at laposte.net  Thu Mar  5 23:51:31 2009
From: kawarod at laposte.net (rod)
Date: Fri, 06 Mar 2009 11:51:31 +0400
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com>
References: <49AFC1C3.9030603@laposte.net>	<165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>	<49AFC9BB.9090106@laposte.net>
	<49AFCD02.2000603@laposte.net>
	<9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com>
Message-ID: <49B0D603.502@laposte.net>

using these functions like this did nothing on the SIP INVITE packet :'(

seven wrote:
> try  
> bridge 
> ({effective_caller_id_name 
> ="your_name",effective_caller_id_number="0000"}sofia/b-leg)
>
> On Mar 5, 2009, at 9:00 PM, rod wrote:
>
>   
>> the A leg invite looks like this:
>> From: "Anonymous"<sip:anonymous at anonymous.invalid;user=phone>
>>
>> it has been rewritten like this:
>> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>
>>
>> rod
>>
>> rod wrote:
>>     
>>> Hi Brian,
>>>
>>> if I use the function effective_caller_id_number with my INVITE, I  
>>> get this:
>>>
>>> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS.
>>>
>>> this is not exactly what I'm looking for :p
>>>
>>> rod
>>>
>>>
>>> Brian West wrote:
>>>
>>>       
>>>> Well this depends on how you're placing the call.. if its a standard
>>>> bridge you can on the A-Leg set
>>>> "effective_caller_id_number=000${caller_id_number}" before you call
>>>> bridge.
>>>>
>>>> Is the from already in the correct format?
>>>>
>>>> /b
>>>>
>>>> On Mar 5, 2009, at 6:12 AM, rod wrote:
>>>>
>>>>
>>>>         
>>>>> Dear list,
>>>>>
>>>>> I'd like to rewrite the number in the Remote Party ID header and  
>>>>> only in
>>>>> this header.
>>>>>
>>>>> ex: I'd like to prefix the caller  ID with a prefix code (000 in  
>>>>> this
>>>>> example) in the  RPID header :
>>>>>
>>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>       Remote-Party-ID:
>>>>> <sip: 
>>>>> 123456 
>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>
>>>>> should become:
>>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>       Remote-Party-ID:
>>>>> <sip: 
>>>>> 000123456 
>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>
>>>>> But the From field should remain unchanged.
>>>>>
>>>>> And how to strip this prefix:
>>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>       Remote-Party-ID:
>>>>> <sip: 
>>>>> 000123456 
>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>
>>>>> should become:
>>>>>       From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>       Remote-Party-ID:
>>>>> <sip: 
>>>>> 123456 
>>>>> @10.10.10.10:5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>
>>>>>
>>>>> regards.
>>>>>
>>>>>           
>>>> ------------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>       
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 mashudiflexi at telkom.co.id  Fri Mar  6 00:18:01 2009
From: mashudiflexi at telkom.co.id (mashudi)
Date: Fri, 06 Mar 2009 15:18:01 +0700
Subject: [Freeswitch-users] make freeswitch-snapshot
In-Reply-To: <11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com>
References: <49B0CA95.5080101@telkom.co.id>	<20090306070036.GA24314@jdc.jasonjgw.net>
	<11990ade0903052303s21726d9qa956c52d52418585@mail.gmail.com>
Message-ID: <49B0DC39.1070203@telkom.co.id>

Yes, it works , I would like to say thank you to Stephen Crosby & Jason 
White.

Stephen Crosby wrote:
> I think you need to install the debian package 'unixodbc-dev'
>
> --Stephen
>
> On Thu, Mar 5, 2009 at 11:00 PM, Jason White <jason at jasonjgw.net> wrote:
>   
>> mashudi <mashudiflexi at telkom.co.id> wrote:
>>     
>>> i got error while conduct  ./make  freeswitch-snapshot on debian 2.6 x86
>>> here is the error :
>>>
>>> /usr/bin/ld: cannot find -lodbc
>>>       
>> Have you installed the ODBC library and its development headers? Are they the
>> latest version?
>>
>> It's failing to find the ODBC library.
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> *****************************************
> Sekarang Gratis Nelpon SLJJ Flexi diperluas   ke  
> Yogya
> *****************************************


*****************************************
Sekarang Gratis Nelpon SLJJ Flexi diperluas  ke
Yogya
*****************************************


From gmaruzz at celliax.org  Fri Mar  6 01:39:30 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 6 Mar 2009 10:39:30 +0100
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
	<87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
Message-ID: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>

On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins <msc at freeswitch.org> wrote:
>> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos
>> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t
>> want that.
>
> Repeat the mantra: CentOS is boring and predictable; boring and
> predictable is perfect for real-time telephony systems.
>
>> Any reason why I should not be running Centos with FS? (I do plan on running
>> 64 bit in future though)
>
> None that I can think of unless you have a super cool Linux distro
> that none of us have ever heard of.
>

Maybe, but just maybe, on CentOS you can have a problem running
skypiax (the skype endpoint/trunk): after a couple days of inactivity
the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go
into ininterruptable sleep, causing the Skype clients to go into that
state (the state seen as "D" in top). But I'm not sure about this,
maybe will not be confirmed, needs more investigation. The Jira I
filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27

I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and
CentOS 5.2. BTW: since roughly one month, when the sqlite assert was
fixed, the build on Windows Vista seems rock solid to me.


From nik.middleton at noblesolutions.co.uk  Fri Mar  6 03:04:25 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Fri, 6 Mar 2009 11:04:25 -0000
Subject: [Freeswitch-users] Setting External IP
Message-ID: <A2B4F48489739E439D8845296A2FD9C408BC77@sun.Noble.local>

Hi Guys,

 

In External.xml in  sip profiles I have

 

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

<param name="ext-sip-ip" value="$${external_sip_ip}"/>

 

Can I override these for a given gateway profile?  I have one gateway
that's expecting a local routed IP address due to the way that it's
routed, but the other one expects the public IP, hence the need to make
it gateway specific

 

Regards,

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From Claudio.Cavalera at italtel.it  Fri Mar  6 03:21:41 2009
From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi)
Date: Fri, 6 Mar 2009 12:21:41 +0100
Subject: [Freeswitch-users] 16 threads didn't exit
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0969731@BESONE.corp.dom>
Message-ID: <A2167AE7633A5C4DAE56EEBD1428ECC09697E5@BESONE.corp.dom>

freeswitch-users-bounces at lists.freeswitch.org wrote:
> Hello list,
> I'm trying to track down a seg fault issue with a fs Revision: 11489
> Here is the backtrace pastebin:
> http://pastebin.freeswitch.org/7009
> 
> but before digging the dump I would like to understand: am I the only
> one having error like this in fs console:
> "Error in my_thread_global_end(): 16 threads didn't exit"
> 
> I'm asking this because googling around did not take me to
> much relation
> between this error and fs.
> In fact as you can see the error does not have the usual fs logging
> format with date time and logging level, it's just a yellow
> line printed
> out in console.


Hello,
I'm trying to track down the source of this "problem".
For this reason I would like to redirect this message to a log file so
that it could be compared and correlated with other logs.
I'm starting fs with this command in a script:
bin/freeswitch -nc -core -log /var/log/freeswitch -conf
/usr/local/freeswitch/conf -db /usr/local/freeswitch/db >>
/var/log/freeswitch/fs_redirection.log 2>>
/var/log/freeswitch/fs_redirection.log

do you think I'm safe and I will capture the error message or the -nc
option could change the behaviour?

Thanks,
Claudio


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From saeedahmad1981 at gmail.com  Fri Mar  6 03:31:35 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Fri, 6 Mar 2009 12:31:35 +0100
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com><8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com><20090306010124.GA12699@jdc.jasonjgw.net><65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
	<8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>
Message-ID: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>

We need a poll.

a) List
b) Forum 

> (b) 

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From helmut.kuper at ewetel.de  Fri Mar  6 03:31:16 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 06 Mar 2009 12:31:16 +0100
Subject: [Freeswitch-users] Setting External IP
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC77@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC77@sun.Noble.local>
Message-ID: <49B10984.8080807@ewetel.de>

Hi Nik,

yes you can!

regards
Helmut

On 06.03.2009 12:04, Nik Middleton wrote:
>
> Hi Guys,
>
>  
>
> In External.xml in  sip profiles I have
>
>  
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
>  
>
> Can I override these for a given gateway profile?  I have one gateway
> that?s expecting a local routed IP address due to the way that it?s
> routed, but the other one expects the public IP, hence the need to
> make it gateway specific
>
>  
>
> Regards,
>

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From dujinfang at gmail.com  Fri Mar  6 04:34:10 2009
From: dujinfang at gmail.com (dujinfang)
Date: Fri, 6 Mar 2009 20:34:10 +0800
Subject: [Freeswitch-users] Rewriting Remote Party ID
In-Reply-To: <49B0D603.502@laposte.net>
References: <49AFC1C3.9030603@laposte.net>	<165B6AF4-7F0A-44C8-A6A3-6E67B124EF09@freeswitch.org>	<49AFC9BB.9090106@laposte.net>
	<49AFCD02.2000603@laposte.net>
	<9B1C8998-69AF-4AF2-A21E-FEB5DB43A104@gmail.com>
	<49B0D603.502@laposte.net>
Message-ID: <46348D07-D227-42A5-A25D-A047CE5B1B63@gmail.com>

How about this?

bridge
({origination_caller_id_name
="your_name",origination_caller_id_number="0000"}sofia/b-leg)


On Mar 6, 2009, at 3:51 PM, rod wrote:

> using these functions like this did nothing on the SIP INVITE  
> packet :'(
>
> seven wrote:
>> try
>> bridge
>> ({effective_caller_id_name
>> ="your_name",effective_caller_id_number="0000"}sofia/b-leg)
>>
>> On Mar 5, 2009, at 9:00 PM, rod wrote:
>>
>>
>>> the A leg invite looks like this:
>>> From: "Anonymous"<sip:anonymous at anonymous.invalid;user=phone>
>>>
>>> it has been rewritten like this:
>>> From: "Anonymous" <sip:000000anonymous at 172.29.0.5>
>>>
>>> rod
>>>
>>> rod wrote:
>>>
>>>> Hi Brian,
>>>>
>>>> if I use the function effective_caller_id_number with my INVITE, I
>>>> get this:
>>>>
>>>> From: "Anonymous" <sip: 
>>>> 000000anonymous at 172.29.0.5>;tag=17geyFjX5p0gS.
>>>>
>>>> this is not exactly what I'm looking for :p
>>>>
>>>> rod
>>>>
>>>>
>>>> Brian West wrote:
>>>>
>>>>
>>>>> Well this depends on how you're placing the call.. if its a  
>>>>> standard
>>>>> bridge you can on the A-Leg set
>>>>> "effective_caller_id_number=000${caller_id_number}" before you  
>>>>> call
>>>>> bridge.
>>>>>
>>>>> Is the from already in the correct format?
>>>>>
>>>>> /b
>>>>>
>>>>> On Mar 5, 2009, at 6:12 AM, rod wrote:
>>>>>
>>>>>
>>>>>
>>>>>> Dear list,
>>>>>>
>>>>>> I'd like to rewrite the number in the Remote Party ID header and
>>>>>> only in
>>>>>> this header.
>>>>>>
>>>>>> ex: I'd like to prefix the caller  ID with a prefix code (000 in
>>>>>> this
>>>>>> example) in the  RPID header :
>>>>>>
>>>>>>      From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>>      Remote-Party-ID:
>>>>>> <sip:
>>>>>> 123456
>>>>>> @10.10.10.10 
>>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>>
>>>>>> should become:
>>>>>>      From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>>      Remote-Party-ID:
>>>>>> <sip:
>>>>>> 000123456
>>>>>> @10.10.10.10 
>>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>>
>>>>>> But the From field should remain unchanged.
>>>>>>
>>>>>> And how to strip this prefix:
>>>>>>      From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>>      Remote-Party-ID:
>>>>>> <sip:
>>>>>> 000123456
>>>>>> @10.10.10.10 
>>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>>
>>>>>> should become:
>>>>>>      From: Anonymous<sip:anonymous at anonymous.invalid>;tag=1208367
>>>>>>      Remote-Party-ID:
>>>>>> <sip:
>>>>>> 123456
>>>>>> @10.10.10.10 
>>>>>> :5062;user=phone>;privacy=full;screen=yes;party=calling
>>>>>>
>>>>>>
>>>>>> regards.
>>>>>>
>>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> Freeswitch-users mailing list
>>>> Freeswitch-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 pablosaro at gmail.com  Fri Mar  6 04:54:42 2009
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Fri, 6 Mar 2009 10:54:42 -0200
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
	<87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
	<7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>
Message-ID: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com>

Hi guys,

I'm using Bluewhite64 for my Linux Servers. No problems compiling and
using FS, but not funny if you have dependencies problems (no yum or
aptitude available to solve your problems). Really stable and secure
Linux 64 bit distribution. Obviously, as the system administrator, you
have to take care of keeping the system up to date with security
patches and proper configurations.
Regards

Pablo


On Fri, Mar 6, 2009 at 7:39 AM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> On Fri, Mar 6, 2009 at 1:52 AM, Michael Collins <msc at freeswitch.org> wrote:
>>> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos
>>> 5.2 which is nice to hear.? Yes I know it?s not bleeding edge, but I don?t
>>> want that.
>>
>> Repeat the mantra: CentOS is boring and predictable; boring and
>> predictable is perfect for real-time telephony systems.
>>
>>> Any reason why I should not be running Centos with FS? (I do plan on running
>>> 64 bit in future though)
>>
>> None that I can think of unless you have a super cool Linux distro
>> that none of us have ever heard of.
>>
>
> Maybe, but just maybe, on CentOS you can have a problem running
> skypiax (the skype endpoint/trunk): after a couple days of inactivity
> the snd-dummy ALSA driver of CentOS (at least on 32 bit) seems to go
> into ininterruptable sleep, causing the Skype clients to go into that
> state (the state seen as "D" in top). But I'm not sure about this,
> maybe will not be confirmed, needs more investigation. The Jira I
> filed for this is: http://jira.freeswitch.org/browse/MODSKYPIAX-27
>
> I had very good overall experiences with Ubuntu 8.04 LTS Hardy, and
> CentOS 5.2. BTW: since roughly one month, when the sqlite assert was
> fixed, the build on Windows Vista seems rock solid to me.
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 edpimentl at gmail.com  Fri Mar  6 05:01:19 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 6 Mar 2009 08:01:19 -0500
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
	<87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
	<7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>
	<247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com>
Message-ID: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com>

Anyone using uBuntu 8.10 and XEN?
What has been your most stable VM / FS platform?
Thanks in advance,
-E
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From sergey.kirillov at gmail.com  Fri Mar  6 05:03:08 2009
From: sergey.kirillov at gmail.com (Sergey Kirillov)
Date: Fri, 06 Mar 2009 15:03:08 +0200
Subject: [Freeswitch-users] OpenZap and Sangoma A500 BRI card
Message-ID: <49B11F0C.6040706@gmail.com>

Hi everybody,

I'm trying to use Sangoma A500 BRI card with OpenZap, but it does not work.

Can somebody help to to configure it?

Problem log (Incoming call):
2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() 
Unhandled event 2
2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() 
Unhandled event 2
2009-03-06 14:58:14 [WARNING] ozmod_wanpipe.c:953 wanpipe_next_event() 
Unhandled event 2
2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 
switch_channel_set_name() New Channel OpenZAP/1:1/2360012 
[7473c92a-0a4e-11de-9dc3-c56d4d411902]
2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() 
Processing 80503820933->2360012 in context default
2009-03-06 14:58:14 [NOTICE] switch_ivr.c:1343 
switch_ivr_session_transfer() Transfer OpenZAP/1:1/2360012 to 
XML[1000 at default]
2009-03-06 14:58:14 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() 
Processing 80503820933->1000 in context default
2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx 
XML features
2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 
record_session::/opt/freeswitch/recordings/80503820933.2009-03-06-14-58-14.wav
2009-03-06 14:58:14 [INFO] switch_ivr_async.c:1730 
switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf 
XML features
2009-03-06 14:58:14 [NOTICE] switch_channel.c:567 
switch_channel_set_name() New Channel 
sofia/internal/sip:1000 at 192.168.122.1:5061;transport=udp 
[748a2ba2-0a4e-11de-9dc3-c56d4d411902]
2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state
2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state
2009-03-06 14:58:14 [NOTICE] switch_ivr_originate.c:1588 
switch_ivr_originate() Pre-Answer OpenZAP/1:1/2360012!
2009-03-06 14:58:14 [ERR] Span:0 Q.921() Received UA frame in invalid state
2009-03-06 14:58:14 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received 
unhandled message 125 (0x7d)
2009-03-06 14:58:15 [CRIT] ozmod_isdn.c:964 zap_isdn_931_34() Received 
unhandled message 125 (0x7d)
2009-03-06 14:58:26 [NOTICE] Span:0 Q.921() I frame in invalid state ignored



Here are my config files

--- openzap.conf --
[span wanpipe BRI_1]
name => BRI_1
trunk_type => bri
b-channel => 1:1-2
d-channel => 1:3


--- openzap.conf.xml ---
<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<pri_spans>
<span name="BRI_1">
<!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
<param name="q921loglevel" value="info"/>
<param name="q931loglevel" value="info"/>
<param name="mode" value="user"/>
<param name="dialect" value="q931"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
</configuration>


--- wanpipe1.conf ---
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
wp1aft1 = wanpipe1, auto, API, Comment
wp1aft2 = wanpipe1, auto, API, Comment

[wanpipe1]
CARD_TYPE         = AFT
S514CPU           = A
CommPort          = PRI
AUTO_PCISLOT      = NO
PCISLOT           = 4
PCIBUS            = 5
FE_MEDIA          = E1
FE_LCODE          = HDB3
FE_FRAME          = CRC4
FE_LINE           = 1
TE_CLOCK          = NORMAL
TE_REF_CLOCK      = 0
TE_HIGHIMPEDANCE  = NO
TE_RX_SLEVEL      = 120
LBO               = 120OH
TE_SIG_MODE       = CCS
FE_TXTRISTATE     = NO
MTU               = 1500
UDPPORT           = 9000
TTL               = 255
IGNORE_FRONT_END  = NO
TDMV_HW_DTMF      = NO

[wp1aft1]
HDLC_STREAMING  = NO
ACTIVE_CH       = 1-15.17-31
IDLE_FLAG       = 0x7E
MTU             = 240
MRU             = 240
DATA_MUX        = NO
TDMV_HWEC       = NO

[wp1aft2]
HDLC_STREAMING  = YES
ACTIVE_CH       = 16
MTU             = 1500
MRU             = 1500
DATA_MUX        = NO
TDMV_HWEC       = NO




From pablosaro at gmail.com  Fri Mar  6 05:41:01 2009
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Fri, 6 Mar 2009 11:41:01 -0200
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
	<87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
	<7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>
	<247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com>
	<9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com>
Message-ID: <247f8100903060541v10ab6605hc1d4e4de52f3db9b@mail.gmail.com>

I've tried FS on ESX, but not cheap. Works good for small
environments. I've never virtualized FS for big or critical systems.

Pablo


On Fri, Mar 6, 2009 at 11:01 AM, EdPimentl <edpimentl at gmail.com> wrote:
> Anyone using uBuntu 8.10 and XEN?
> What has been your most stable VM / FS platform?
> Thanks in advance,
> -E
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 Mar  6 06:00:28 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Mar 2009 08:00:28 -0600
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
Message-ID: <191c3a030903060600v7defb8adi683ffd2ff5b36b3b@mail.gmail.com>

We are considering asking CentOS to make a "FS cut" set of packages ideal
for a telephony server with one install choice.


On Thu, Mar 5, 2009 at 5:59 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Just curious here.
>
>
>
> I?ve always followed the fedora route but became disillusioned with the
> focus on the desktop rather than the server mode.  Of late I?ve moved my
> servers to Centos.  I felt the need for stable systems.
>
>
>
> Everyone seems to slate Centos, but to my surprise Anthony recommends
> Centos 5.2 which is nice to hear.  Yes I know it?s not bleeding edge, but I
> don?t want that.
>
>
>
> Any reason why I should not be running Centos with FS? (I do plan on
> running 64 bit in future though)
>
>
>
> Regards,
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar  6 06:21:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Mar 2009 08:21:08 -0600
Subject: [Freeswitch-users] 16 threads didn't exit
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC09697E5@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0969731@BESONE.corp.dom>
	<A2167AE7633A5C4DAE56EEBD1428ECC09697E5@BESONE.corp.dom>
Message-ID: <191c3a030903060621r2592a89i208a7cc9cf655d91@mail.gmail.com>

I recall someone asking about this before and it has to be one of the
depends.
Do you load anything that is not enabled by default in the standard install.

It says my_ in it could it be mysql ?

Let's google "my_thread_global_end()" and see...

Checking http://www.google.com/search?q=my_thread_global_end()

hey!
http://forums.mysql.com/read.php?10,153077,153077

Not sure if its the same thing or not but it looks pretty similar.




On Fri, Mar 6, 2009 at 5:21 AM, Cavalera Claudio Luigi <
Claudio.Cavalera at italtel.it> wrote:

> freeswitch-users-bounces at lists.freeswitch.org wrote:
> > Hello list,
> > I'm trying to track down a seg fault issue with a fs Revision: 11489
> > Here is the backtrace pastebin:
> > http://pastebin.freeswitch.org/7009
> >
> > but before digging the dump I would like to understand: am I the only
> > one having error like this in fs console:
> > "Error in my_thread_global_end(): 16 threads didn't exit"
> >
> > I'm asking this because googling around did not take me to
> > much relation
> > between this error and fs.
> > In fact as you can see the error does not have the usual fs logging
> > format with date time and logging level, it's just a yellow
> > line printed
> > out in console.
>
>
> Hello,
> I'm trying to track down the source of this "problem".
> For this reason I would like to redirect this message to a log file so
> that it could be compared and correlated with other logs.
> I'm starting fs with this command in a script:
> bin/freeswitch -nc -core -log /var/log/freeswitch -conf
> /usr/local/freeswitch/conf -db /usr/local/freeswitch/db >>
> /var/log/freeswitch/fs_redirection.log 2>>
> /var/log/freeswitch/fs_redirection.log
>
> do you think I'm safe and I will capture the error message or the -nc
> option could change the behaviour?
>
> Thanks,
> Claudio
>
>
> Internet Email Confidentiality Footer
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Fri Mar  6 06:23:23 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Mar 2009 08:23:23 -0600
Subject: [Freeswitch-users] Setting External IP
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C408BC77@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C408BC77@sun.Noble.local>
Message-ID: <191c3a030903060623p16dabe4bw5f4ed1fb2d196ca4@mail.gmail.com>

gateways are children of profiles so if you need them to be separate you
need to make 2 profiles and run the other one on another IP or another port.


On Fri, Mar 6, 2009 at 5:04 AM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> In External.xml in  sip profiles I have
>
>
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>
>
>
> Can I override these for a given gateway profile?  I have one gateway
> that?s expecting a local routed IP address due to the way that it?s routed,
> but the other one expects the public IP, hence the need to make it gateway
> specific
>
>
>
> Regards,
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Fri Mar  6 06:27:27 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 6 Mar 2009 08:27:27 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
	<8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
	<20090306010124.GA12699@jdc.jasonjgw.net>
	<65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
	<8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>
	<5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>
Message-ID: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com>

Did anybody notice my email from yesterday that shows how there already is a
forum on voip-info that is linked to our homepage and nobody uses it?

We can't take this poll until we have a list of volunteers who would manage
any new
online resources.

On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

>   We need a poll.
>
> a) List
> b) Forum
>
> > (b)
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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/

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Mar  6 08:12:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 6 Mar 2009 08:12:37 -0800
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
	<8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
	<20090306010124.GA12699@jdc.jasonjgw.net>
	<65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
	<8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>
	<5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>
	<191c3a030903060627y2102ebccscadf0550f7ba0d57@mail.gmail.com>
Message-ID: <87f2f3b90903060812s13bccf49ma7b31a1dadf5f272@mail.gmail.com>

> Did anybody notice my email from yesterday that shows how there already is a
> forum on voip-info that is linked to our homepage and nobody uses it?
>
> We can't take this poll until we have a list of volunteers who would manage
> any new
> online resources.

Let's "end the torment of this thread" by giving an official request
for the community:
All of those who are willing and able to moderate a forum please stand
up, go log in to voip-info.org and see what needs to be done to make
it a usable forum. If you are willing to commit to managing the forum
please email me at msc at freeswitch.org. There will *NOT* be a forum if
we don't get at least one motivated community volunteer to take care
of it.

Thanks,
MC


From dujinfang at gmail.com  Fri Mar  6 08:22:41 2009
From: dujinfang at gmail.com (dujinfang)
Date: Sat, 7 Mar 2009 00:22:41 +0800
Subject: [Freeswitch-users] Prefered Linux Distro to run FS on
In-Reply-To: <9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C408BC74@sun.Noble.local>
	<87f2f3b90903051652j216e8e20hc96a69d55bf7270e@mail.gmail.com>
	<7b197bef0903060139q3562c4a4t3da92f22fdd66040@mail.gmail.com>
	<247f8100903060454j28be12d4w7eeebfdcd81c9499@mail.gmail.com>
	<9dc4a1670903060501u5f70d839vb63d872e5d029d0e@mail.gmail.com>
Message-ID: <EB678420-BA7C-4888-BCBF-6F925D4D14F6@gmail.com>

We are using ubuntu 8.04 in Xen(also hosted by ubuntu 8.04, Ubuntu  
8.10 is not xen friendly) as our testing server. It works well,  
however we only use that to test our business logic, not press test at  
all.

On Mar 6, 2009, at 9:01 PM, EdPimentl wrote:

> Anyone using uBuntu 8.10 and XEN?
> What has been your most stable VM / FS platform?
> Thanks in advance,
> -E
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at 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 codecomplete at free.fr  Fri Mar  6 08:30:30 2009
From: codecomplete at free.fr (Fred)
Date: Fri, 06 Mar 2009 17:30:30 +0100
Subject: [Freeswitch-users] Lunchbox-type PC as small server?
Message-ID: <7.0.1.0.2.20090306172918.027b0c68@fredshack.com>

Hello

	I'm looking for a small, lunchbox-like PC to build a small-form 
factor CRM server to sell to small companies. Ideally, it should have 
one PCI slot so that I can stick a voice card to connect to an analog 
phone line and run FreeSwitch as well.

I like Asus' EeeBox (www.asus.com/products.aspx?l1=24&l2=165) but it 
doesn't have room for a PCI slot, and I'm concerned about its performance.

I also like stuff from MiniITX (www.mini-itx.com) , but they're a bit 
pricey, and might also not be fast enough to act as a server.

Are there brands/models you think I should look at?

Thank you.



From vikas.sharma711 at gmail.com  Thu Mar  5 22:55:43 2009
From: vikas.sharma711 at gmail.com (Vikas Sharma)
Date: Fri, 6 Mar 2009 12:25:43 +0530
Subject: [Freeswitch-users] About FreeSwitch
Message-ID: <97d1f10a0903052255n3d3281eeuc9d362811f5e7f29@mail.gmail.com>

HI all,
I am new to this.
Can anybody tell me that freeSwitch can be used as PBX indecently?
Can it be integrated with other  pbx  as  a media server?
If yes,  what  features it has as a media server?

Thnax for any help.

-- 
vikas sharma
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From rupa at rupa.com  Fri Mar  6 03:36:59 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 6 Mar 2009 05:36:59 -0600
Subject: [Freeswitch-users] Please end the torment
In-Reply-To: <5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>
References: <2d9149cd0903051239g1485ba34x8c492948e53c1fad@mail.gmail.com>
	<8CB6C0AA24283A0-890-184F@WEBMAIL-DC14.sysops.aol.com>
	<20090306010124.GA12699@jdc.jasonjgw.net>
	<65bd1c9f0903051726r68711478me4b477fd24101900@mail.gmail.com>
	<8CB6C3D7BE57E31-DEC-266E@webmail-da18.sysops.aol.com>
	<5BEC0926E61241D2989AC317AEBA6B79@SaeedLaptop>
Message-ID: <b39273530903060336r6bded822jb1073098e413a08@mail.gmail.com>

a) List

On Fri, Mar 6, 2009 at 5:31 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:

>   We need a poll.
>
> a) List
> b) Forum
>
> > (b)
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From chris at cloudtel.com  Thu Mar  5 22:54:23 2009
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 5 Mar 2009 22:54:23 -0800
Subject: [Freeswitch-users] make freeswitch-snapshot
In-Reply-To: <49B0CA95.5080101@telkom.co.id>
References: <49B0CA95.5080101@telkom.co.id>
Message-ID: <200903052254.23746.chris@cloudtel.com>

apt-get install unixodbc-dev

On March 5, 2009 11:02:45 pm mashudi wrote:
> Hi Folk,
> i got error while conduct  ./make  freeswitch-snapshot on debian 2.6 x86
> here is the error :
>
> /usr/bin/ld: cannot find -lodbc
> collect2: ld returned 1 exit status
> make[2]: *** [libfreeswitch.la] Error 1
> Making all in src
> Making all in mod
>
> making all mod_amr
> make[5]: *** No rule to make target
> `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by
> `mod_amr.so'.  Stop.
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running: